tizen 2.4 releasetizen_2.4_mobile_releasesubmit/tizen_2.4/20151028.064339accepted/tizen/2.4/mobile/20151029.032704accepted/tizen_2.4_mobile

11 files changed, 11384 insertions, 0 deletions

diff --git a/gst/rtsp/Makefile.am b/gst/rtsp/Makefile.am
new file mode 100755
index 0000000..267e49a
--- /dev/null
+++ b/gst/rtsp/Makefile.am
@@ -0,0 +1,16 @@
+plugin_LTLIBRARIES = libgstrtsp.la
+
+libgstrtsp_la_SOURCES = gstrtsp.c gstrtspsrc.c \
+			gstrtpdec.c gstrtspext.c
+
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+libgstrtsp_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) $(GST_LIBS) $(GST_BASE_LIBS) $(GIO_LIBS) \
+		       -lgstrtp-@GST_API_VERSION@ -lgstrtsp-@GST_API_VERSION@ \
+		       -lgstsdp-@GST_API_VERSION@ $(GST_NET_LIBS) $(GST_LIBS)
+libgstrtsp_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstrtsp_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
+
+noinst_HEADERS = gstrtspsrc.h     \
+		 gstrtsp.h        \
+		 gstrtpdec.h      \
+		 gstrtspext.h
diff --git a/gst/rtsp/Makefile.in b/gst/rtsp/Makefile.in
new file mode 100755
index 0000000..34928f7
--- /dev/null
+++ b/gst/rtsp/Makefile.in
@@ -0,0 +1,928 @@
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+	    || cp -p $$d/$$file "$(distdir)/$$file" \
+	    || exit 1; \
+	  fi; \
+	done
+check-am: all-am
+check: check-am
+all-am: Makefile $(LTLIBRARIES) $(HEADERS)
+installdirs:
+	for dir in "$(DESTDIR)$(plugindir)"; do \
+	  test -z "$$dir" || $(MKDIR_P) "$$dir"; \
+	done
+install: install-am
+install-exec: install-exec-am
+install-data: install-data-am
+uninstall: uninstall-am
+
+install-am: all-am
+	@$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
+
+installcheck: installcheck-am
+install-strip:
+	if test -z '$(STRIP)'; then \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	      install; \
+	else \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	    "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'" install; \
+	fi
+mostlyclean-generic:
+
+clean-generic:
+
+distclean-generic:
+	-test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+	-test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
+
+maintainer-clean-generic:
+	@echo "This command is intended for maintainers to use"
+	@echo "it deletes files that may require special tools to rebuild."
+clean: clean-am
+
+clean-am: clean-generic clean-libtool clean-pluginLTLIBRARIES \
+	mostlyclean-am
+
+distclean: distclean-am
+	-rm -rf ./$(DEPDIR)
+	-rm -f Makefile
+distclean-am: clean-am distclean-compile distclean-generic \
+	distclean-tags
+
+dvi: dvi-am
+
+dvi-am:
+
+html: html-am
+
+html-am:
+
+info: info-am
+
+info-am:
+
+install-data-am: install-pluginLTLIBRARIES
+
+install-dvi: install-dvi-am
+
+install-dvi-am:
+
+install-exec-am:
+
+install-html: install-html-am
+
+install-html-am:
+
+install-info: install-info-am
+
+install-info-am:
+
+install-man:
+
+install-pdf: install-pdf-am
+
+install-pdf-am:
+
+install-ps: install-ps-am
+
+install-ps-am:
+
+installcheck-am:
+
+maintainer-clean: maintainer-clean-am
+	-rm -rf ./$(DEPDIR)
+	-rm -f Makefile
+maintainer-clean-am: distclean-am maintainer-clean-generic
+
+mostlyclean: mostlyclean-am
+
+mostlyclean-am: mostlyclean-compile mostlyclean-generic \
+	mostlyclean-libtool
+
+pdf: pdf-am
+
+pdf-am:
+
+ps: ps-am
+
+ps-am:
+
+uninstall-am: uninstall-pluginLTLIBRARIES
+
+.MAKE: install-am install-strip
+
+.PHONY: CTAGS GTAGS TAGS all all-am check check-am clean clean-generic \
+	clean-libtool clean-pluginLTLIBRARIES cscopelist-am ctags \
+	ctags-am distclean distclean-compile distclean-generic \
+	distclean-libtool distclean-tags distdir dvi dvi-am html \
+	html-am info info-am install install-am install-data \
+	install-data-am install-dvi install-dvi-am install-exec \
+	install-exec-am install-html install-html-am install-info \
+	install-info-am install-man install-pdf install-pdf-am \
+	install-pluginLTLIBRARIES install-ps install-ps-am \
+	install-strip installcheck installcheck-am installdirs \
+	maintainer-clean maintainer-clean-generic mostlyclean \
+	mostlyclean-compile mostlyclean-generic mostlyclean-libtool \
+	pdf pdf-am ps ps-am tags tags-am uninstall uninstall-am \
+	uninstall-pluginLTLIBRARIES
+
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:
diff --git a/gst/rtsp/README b/gst/rtsp/README
new file mode 100755
index 0000000..0245891
--- /dev/null
+++ b/gst/rtsp/README
@@ -0,0 +1,377 @@
+RTSP source 
+-----------
+
+The RTSP source establishes a connection to an RTSP server and sets up
+the UDP sources and RTP session handlers.
+
+An RTSP session is created as follows:
+
+- Parse RTSP URL:
+
+   ex:
+     rtsp://thread:5454/south-rtsp.mp3
+
+- Open a connection to the server with the url. All further conversation with
+  the server should be done with this connection. Each request/reply has
+  a CSeq number added to the header.
+
+- Send a DESCRIBE request for the url. We currently support a response in
+  SDP.
+
+  ex:
+
+    >> DESCRIBE rtsp://thread:5454/south-rtsp.mp3 RTSP/1.0
+    >> Accept: application/sdp
+    >> CSeq: 0
+    >>
+    << RTSP/1.0 200 OK
+    << Content-Length: 84
+    << Content-Type: application/sdp
+    << CSeq: 0
+    << Date: Wed May 11 13:09:37 2005 GMT
+    <<
+    << v=0
+    << o=- 0 0 IN IP4 192.168.1.1
+    << s=No Title
+    << m=audio 0 RTP/AVP 14
+    << a=control:streamid=0
+
+- Parse the SDP message, for each stream (m=) we create an GstRTPStream. We need
+  to allocate two local UDP ports for receiving the RTP and RTCP data because we
+  need to send the port numbers to the server in the next request.
+
+  In RTSPSrc we first create an element that can handle the udp://0.0.0.0:0 uri. This
+  will create an udp source element with a random port number. We get the port
+  number by getting the "port" property of the element after setting the element to 
+  PAUSED. This element is used for the RTP packets and has to be an even number. If
+  the random port number is not an even number we retry to allocate a new udp source.
+
+  We then create another UDP source element with the next (uneven) port number to 
+  receive the RTCP packet on. After this step we have two udp ports we can use to
+  accept RTP packets.
+
+    +-----------------+
+    | +------------+  |
+    | | udpsrc0    |  |
+    | |  port=5000 |  |
+    | +------------+  |
+    | +------------+  |
+    | | udpsrc1    |  |
+    | |  port=5001 |  |
+    | +------------+  |
+    +-----------------+
+
+- Send a SETUP message to the server with the RTP ports. We get the setup URI from
+  the a= attribute in the SDP message. This can be an absolute URL or a relative
+  url.
+
+  ex:
+
+    >> SETUP rtsp://thread:5454/south-rtsp.mp3/streamid=0 RTSP/1.0
+    >> CSeq: 1
+    >> Transport: RTP/AVP/UDP;unicast;client_port=5000-5001,RTP/AVP/UDP;multicast,RTP/AVP/TCP
+    >>
+    << RTSP/1.0 200 OK
+    << Transport: RTP/AVP/UDP;unicast;client_port=5000-5001;server_port=6000-6001
+    << CSeq: 1
+    << Date: Wed May 11 13:21:43 2005 GMT
+    << Session: 5d5cb94413288ccd
+    <<
+
+  The client needs to send the local ports to the server along with the supported 
+  transport types. The server selects the final transport which it returns in the
+  Transport header field. The server also includes its ports where RTP and RTCP
+  messages can be sent to.
+
+  In the above example UDP was choosen as a transport. At this point the RTSPSrc element
+  will furter configure its elements to process this stream.
+
+  The RTSPSrc will create and connect an RTP session manager element and will
+  connect it to the src pads of the udp element. The data pad from the RTP session 
+  manager is ghostpadded to RTPSrc.
+  The RTCP pad of the rtpdec is routed to a new udpsink that sends data to the RTCP
+  port of the server as returned in the Transport: header field.
+
+    +---------------------------------------------+
+    | +------------+                              |
+    | | udpsrc0    |   +--------+                 |
+    | |  port=5000 ----- rtpdec --------------------
+    | +------------+   |        |                 |
+    | +------------+   |        |  +------------+ |
+    | | udpsrc1    ----- RTCP   ---- udpsink    | |
+    | |  port=5001 |   +--------+  |  port=6001 | |
+    | +------------+               +------------+ |
+    +---------------------------------------------+
+
+  The output type of rtpdec is configured as the media type specified in the SDP
+  message. 
+
+- All the elements are set to PAUSED/PLAYING and the PLAY RTSP message is
+  sent.
+
+    >> PLAY rtsp://thread:5454/south-rtsp.mp3 RTSP/1.0
+    >> CSeq: 2
+    >> Session: 5d5cb94413288ccd
+    >>
+    << RTSP/1.0 200 OK
+    << CSeq: 2
+    << Date: Wed May 11 13:21:43 2005 GMT
+    << Session: 5d5cb94413288ccd
+    <<
+
+- The udp source elements receive data from that point and the RTP/RTCP messages
+  are processed by the elements.
+
+- In the case of interleaved mode, the SETUP method yields:
+
+    >> SETUP rtsp://thread:5454/south-rtsp.mp3/streamid=0 RTSP/1.0
+    >> CSeq: 1
+    >> Transport: RTP/AVP/UDP;unicast;client_port=5000-5001,RTP/AVP/UDP;multicast,RTP/AVP/TCP
+    >>
+    << RTSP/1.0 200 OK
+    << Transport: RTP/AVP/TCP;interleaved=0-1
+    << CSeq: 1
+    << Date: Wed May 11 13:21:43 2005 GMT
+    << Session: 5d5cb94413288ccd
+    <<
+
+  This means that RTP/RTCP messages will be sent on channel 0/1 respectively and that
+  the data will be received on the same connection as the RTSP connection.
+
+  At this point, we remove the UDP source elements as we don't need them anymore. We
+  set up the rtpsess session manager element though as follows:
+
+    +---------------------------------------------+
+    | +------------+                              |
+    | | _loop()    |   +--------+                 |
+    | |            ----- rtpses --------------------
+    | |            |   |        |                 |
+    | |            |   |        |  +------------+ |
+    | |            ----- RTCP   ---- udpsink    | |
+    | |            |   +--------+  |  port=6001 | |
+    | +------------+               +------------+ |
+    +---------------------------------------------+
+
+  We start an interal task to start reading from the RTSP connection waiting
+  for data. The received data is then pushed to the rtpdec element.
+
+  When reading from the RTSP connection we receive data packets in the
+  following layout (see also RFC2326)
+
+    $<1 byte channel><2 bytes length, big endian><length bytes of data>
+
+
+RTSP server
+-----------
+
+An RTSP server listen on a port (default 554) for client connections. The client
+typically keeps this channel open during the RTSP session to instruct the server
+to pause/play/stop the stream.
+
+The server exposes a stream consisting of one or more media streams using an
+URL. The media streams are typically audio and video.
+
+  ex:
+     rtsp://thread:5454/alien-rtsp.mpeg
+
+     exposes an audio/video stream. The video is mpeg packetized in RTP and
+     the audio is mp3 in RTP.
+
+The streaming server typically uses a different channel to send the media
+data to clients, typically using RTP over UDP. It is also possible to stream
+the data to the client using the initial RTSP TCP session (the interleaved
+mode). This last mode is usufull when the client is behind a firewall but
+does not take advantage of the RTP/UDP features.
+
+In both cases, media data is send to the clients in an unmultiplexed format
+packetized as RTP packets.
+
+The streaming server has to negotiate a connection protocol for each of the
+media streams with the client.
+
+Minimal server requirements:
+
+- The server should copy the CSeq header field in a client request to the
+  response so that the client can match the response to the request.
+
+- The server should keep a session for each client after the client issued
+  a SETUP command. The client should use the same session id for all future
+  request to this server.
+
+- the server must support an OPTIONS request send over the RTSP connection.
+
+    >> OPTIONS * RTSP/1.0
+    >> CSeq: 1
+    >>
+    << RTSP/1.0 200 OK
+    << CSeq: 1
+    << Date: Wed May 11 13:21:43 2005 GMT
+    << Session: 5d5cb94413288ccd
+    << Public: DESCRIBE, SETUP, TEARDOWN, PLAY
+    <<
+
+   The OPTIONS request should list all supported methods on the server.
+ 
+ - The server should support the DESCRIBE method.
+
+    >> DESCRIBE rtsp://thread:5454/south-rtsp.mp3 RTSP/1.0
+    >> Accept: application/sdp
+    >> CSeq: 2
+    >>
+    << RTSP/1.0 200 OK
+    << Content-Length: 84
+    << Content-Type: application/sdp
+    << CSeq: 2
+    << Date: Wed May 11 13:09:37 2005 GMT
+    <<
+    << v=0
+    << o=- 0 0 IN IP4 192.168.1.1
+    << s=No Title
+    << m=audio 0 RTP/AVP 14
+    << a=control:streamid=0
+
+    The client issues a DESCRIBE command for a specific URL that corresponds
+    to an available stream. The client will also send an Accept header to
+    list its supported formats. 
+
+    The server answers this request with a reply in one of the client supported
+    formats (application/sdp is the most common). The server typically sends a
+    fixed reply to all clients for each configured stream.
+
+ - The server must support the SETUP command to configure the media streams 
+   that were listed in the DESCRIBE command.
+
+    >> SETUP rtsp://thread:5454/south-rtsp.mp3/streamid=0 RTSP/1.0
+    >> CSeq: 3
+    >> Transport: RTP/AVP/UDP;unicast;client_port=5000-5001,RTP/AVP/UDP;multicast,RTP/AVP/TCP
+    >>
+    << RTSP/1.0 200 OK
+    << Transport: RTP/AVP/UDP;unicast;client_port=5000-5001;server_port=6000-6001
+    << CSeq: 3
+    << Date: Wed May 11 13:21:43 2005 GMT
+    << Session: 5d5cb94413288ccd
+
+    The client will send a SETUP command for each of the streams listed in the 
+    DESCRIBE reply. For sdp will use a URL as listed in the a=control: property.
+
+    The client will list the supported transports in the Transport: header field.
+    Each transport is separated with a comma (,) and listed in order of preference.
+    The server has to select the first supported transport.
+ 
+    In the above example 3 transport are listed:
+
+       RTP/AVP/UDP;unicast;client_port=5000-5001
+
+         The client will accept RTP over UDP on the port pair 5000-5001. Port
+	 5000 will accept the RTP packets, 5001 the RTSP packets send by the
+	 server.
+       
+       RTP/AVP/UDP;multicast
+
+         The client can join a multicast group for the specific media stream.
+	 The port numbers of the multicast group it will connect to have to
+	 be specified by the server in the reply.
+       
+       RTP/AVP/TCP
+
+         the client can accept RTP packets interleaved on the RTSP connection.
+
+    The server selects a supported transport an allocates an RTP port pair to
+    receive RTP and RTSP data from the client. This last step is optional when
+    the server does not accept RTP data.
+
+    The server should allocate a session for the client and should send the
+    sessionId to the client. The client should use this session id for all
+    future requests.
+
+    The server may refuse a client that does not use the same transport method
+    for all media streams.
+
+    The server stores all client port pairs in the server client session along
+    with the transport method.
+
+    ex:
+
+      For an on-demand stream the server could construct a (minimal) RTP GStreamer 
+      pipeline for the client as follows (for an mp3 stream):
+    
+      +---------+    +-----------+    +------------+    +-------------+
+      | filesrc |    | rtpmp3enc |    | rtpsession |    | udpsink     |
+      |         |    |           |    |            |    |   host=XXX  |
+      |         |    |           |    |            |    |   port=5000 |
+      |        src--sink        src--rtpsink   rtpsrc--sink           |
+      +---------+    +-----------+    |            |    +-------------+
+                                      |            |    +-------------+
+                                      |            |    | udpsink     |
+                                      |            |    |   host=XXX  |
+                                      |            |    |   port=5001 |
+                                      |       rtspsrc--sink           |
+                                      +------------+    +-------------+
+
+      The server would set the above pipeline to PAUSE to make sure no data
+      is sent to the client yet.
+
+      optionally udpsrc elements can be configured to receive client RTP and
+      RTSP messages.
+
+    ex:
+
+      For a live stream the server could construct a (minimal) RTP GStreamer 
+      pipeline for the clients as follows (for an mp3 stream):
+    
+      +---------+    +--------+    +-----------+    +------------+    +--------------+
+      | source  |    | mp3enc |    | rtpmp3enc |    | rtpsession |    | multiudpsink |
+      |         |    |        |    |           |    |            |    |   host=...   |
+      |         |    |        |    |           |    |            |    |   port=...   |
+      |        src--sink     src--sink        src--rtpsink   rtpsrc--sink            |
+      +---------+    +--------+    +-----------+    |            |    +--------------+
+                                                    |            |    +--------------+
+                                                    |            |    | multiudpsink |
+                                                    |            |    |   host=...   |
+                                                    |            |    |   port=...   |
+                                                    |       rtspsrc--sink            |
+                                                    +------------+    +--------------+
+
+      Media data is streamed to clients by adding the client host and port numbers
+      to the multiudpsinks.
+
+      optionally udpsrc elements can be configured to receive client RTP and
+      RTSP messages.
+     
+ - The server must support the PLAY command to start playback of the configured
+   media streams.
+
+    >> PLAY rtsp://thread:5454/south-rtsp.mp3 RTSP/1.0
+    >> CSeq: 2
+    >> Session: 5d5cb94413288ccd
+    >>
+    << RTSP/1.0 200 OK
+    << CSeq: 2
+    << Date: Wed May 11 13:21:43 2005 GMT
+    << Session: 5d5cb94413288ccd
+    <<
+
+    Using the Session: header field, the server finds the pipeline of the session
+    to PLAY and sets the pipeline to PLAYING at which point the client receives
+    the media stream data.
+
+    In case of a live stream, the server adds the port numbers to a multiudpsink
+    element.
+
+ - The server must support the TEARDOWN command to stop playback and free the
+   session of a client.
+
+    >> TEARDOWN rtsp://thread:5454/south-rtsp.mp3 RTSP/1.0
+    >> CSeq: 4
+    >> Session: 5d5cb94413288ccd
+    >>
+    << RTSP/1.0 200 OK
+    << CSeq: 4
+    << Date: Wed May 11 13:21:43 2005 GMT
+    <<
+
+    The server destroys the client pipeline in case of an on-demand stream or
+    removes the client ports from the multiudpsinks. This effectively stops
+    streaming to the client.
+
+
diff --git a/gst/rtsp/gstrtpdec.c b/gst/rtsp/gstrtpdec.c
new file mode 100755
index 0000000..e24927b
--- /dev/null
+++ b/gst/rtsp/gstrtpdec.c
@@ -0,0 +1,897 @@
+/* GStreamer
+ * Copyright (C) <2005,2006> Wim Taymans <wim.taymans@gmail.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+/* Element-Checklist-Version: 5 */
+
+/**
+ * SECTION:element-rtpdec
+ *
+ * A simple RTP session manager used internally by rtspsrc.
+ */
+
+/* #define HAVE_RTCP */
+
+#include <gst/rtp/gstrtpbuffer.h>
+
+#ifdef HAVE_RTCP
+#include <gst/rtp/gstrtcpbuffer.h>
+#endif
+
+#include "gstrtpdec.h"
+#include <stdio.h>
+
+GST_DEBUG_CATEGORY_STATIC (rtpdec_debug);
+#define GST_CAT_DEFAULT (rtpdec_debug)
+
+/* GstRTPDec signals and args */
+enum
+{
+  SIGNAL_REQUEST_PT_MAP,
+  SIGNAL_CLEAR_PT_MAP,
+
+  SIGNAL_ON_NEW_SSRC,
+  SIGNAL_ON_SSRC_COLLISION,
+  SIGNAL_ON_SSRC_VALIDATED,
+  SIGNAL_ON_BYE_SSRC,
+  SIGNAL_ON_BYE_TIMEOUT,
+  SIGNAL_ON_TIMEOUT,
+  LAST_SIGNAL
+};
+
+#define DEFAULT_LATENCY_MS      200
+
+enum
+{
+  PROP_0,
+  PROP_LATENCY
+};
+
+static GstStaticPadTemplate gst_rtp_dec_recv_rtp_sink_template =
+GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%u",
+    GST_PAD_SINK,
+    GST_PAD_REQUEST,
+    GST_STATIC_CAPS ("application/x-rtp")
+    );
+
+static GstStaticPadTemplate gst_rtp_dec_recv_rtcp_sink_template =
+GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%u",
+    GST_PAD_SINK,
+    GST_PAD_REQUEST,
+    GST_STATIC_CAPS ("application/x-rtcp")
+    );
+
+static GstStaticPadTemplate gst_rtp_dec_recv_rtp_src_template =
+GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%u_%u_%u",
+    GST_PAD_SRC,
+    GST_PAD_SOMETIMES,
+    GST_STATIC_CAPS ("application/x-rtp")
+    );
+
+static GstStaticPadTemplate gst_rtp_dec_rtcp_src_template =
+GST_STATIC_PAD_TEMPLATE ("rtcp_src_%u",
+    GST_PAD_SRC,
+    GST_PAD_REQUEST,
+    GST_STATIC_CAPS ("application/x-rtcp")
+    );
+
+static void gst_rtp_dec_finalize (GObject * object);
+static void gst_rtp_dec_set_property (GObject * object,
+    guint prop_id, const GValue * value, GParamSpec * pspec);
+static void gst_rtp_dec_get_property (GObject * object,
+    guint prop_id, GValue * value, GParamSpec * pspec);
+
+static GstClock *gst_rtp_dec_provide_clock (GstElement * element);
+static GstStateChangeReturn gst_rtp_dec_change_state (GstElement * element,
+    GstStateChange transition);
+static GstPad *gst_rtp_dec_request_new_pad (GstElement * element,
+    GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
+static void gst_rtp_dec_release_pad (GstElement * element, GstPad * pad);
+
+static GstFlowReturn gst_rtp_dec_chain_rtp (GstPad * pad, GstObject * parent,
+    GstBuffer * buffer);
+static GstFlowReturn gst_rtp_dec_chain_rtcp (GstPad * pad, GstObject * parent,
+    GstBuffer * buffer);
+
+
+/* Manages the receiving end of the packets.
+ *
+ * There is one such structure for each RTP session (audio/video/...).
+ * We get the RTP/RTCP packets and stuff them into the session manager. 
+ */
+struct _GstRTPDecSession
+{
+  /* session id */
+  gint id;
+  /* the parent bin */
+  GstRTPDec *dec;
+
+  gboolean active;
+  /* we only support one ssrc and one pt */
+  guint32 ssrc;
+  guint8 pt;
+  GstCaps *caps;
+
+  /* the pads of the session */
+  GstPad *recv_rtp_sink;
+  GstPad *recv_rtp_src;
+  GstPad *recv_rtcp_sink;
+  GstPad *rtcp_src;
+};
+
+/* find a session with the given id */
+static GstRTPDecSession *
+find_session_by_id (GstRTPDec * rtpdec, gint id)
+{
+  GSList *walk;
+
+  for (walk = rtpdec->sessions; walk; walk = g_slist_next (walk)) {
+    GstRTPDecSession *sess = (GstRTPDecSession *) walk->data;
+
+    if (sess->id == id)
+      return sess;
+  }
+  return NULL;
+}
+
+/* create a session with the given id */
+static GstRTPDecSession *
+create_session (GstRTPDec * rtpdec, gint id)
+{
+  GstRTPDecSession *sess;
+
+  sess = g_new0 (GstRTPDecSession, 1);
+  sess->id = id;
+  sess->dec = rtpdec;
+  rtpdec->sessions = g_slist_prepend (rtpdec->sessions, sess);
+
+  return sess;
+}
+
+static void
+free_session (GstRTPDecSession * session)
+{
+  g_free (session);
+}
+
+static guint gst_rtp_dec_signals[LAST_SIGNAL] = { 0 };
+
+#define gst_rtp_dec_parent_class parent_class
+G_DEFINE_TYPE (GstRTPDec, gst_rtp_dec, GST_TYPE_ELEMENT);
+
+static void
+gst_rtp_dec_class_init (GstRTPDecClass * g_class)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstRTPDecClass *klass;
+
+  klass = (GstRTPDecClass *) g_class;
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+
+  GST_DEBUG_CATEGORY_INIT (rtpdec_debug, "rtpdec", 0, "RTP decoder");
+
+  gobject_class->finalize = gst_rtp_dec_finalize;
+  gobject_class->set_property = gst_rtp_dec_set_property;
+  gobject_class->get_property = gst_rtp_dec_get_property;
+
+  g_object_class_install_property (gobject_class, PROP_LATENCY,
+      g_param_spec_uint ("latency", "Buffer latency in ms",
+          "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRTPDec::request-pt-map:
+   * @rtpdec: the object which received the signal
+   * @session: the session
+   * @pt: the pt
+   *
+   * Request the payload type as #GstCaps for @pt in @session.
+   */
+  gst_rtp_dec_signals[SIGNAL_REQUEST_PT_MAP] =
+      g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, request_pt_map),
+      NULL, NULL, g_cclosure_marshal_generic, GST_TYPE_CAPS, 2, G_TYPE_UINT,
+      G_TYPE_UINT);
+
+  gst_rtp_dec_signals[SIGNAL_CLEAR_PT_MAP] =
+      g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, clear_pt_map),
+      NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
+
+  /**
+   * GstRTPDec::on-new-ssrc:
+   * @rtpbin: the object which received the signal
+   * @session: the session
+   * @ssrc: the SSRC 
+   *
+   * Notify of a new SSRC that entered @session.
+   */
+  gst_rtp_dec_signals[SIGNAL_ON_NEW_SSRC] =
+      g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_new_ssrc),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+      G_TYPE_UINT);
+  /**
+   * GstRTPDec::on-ssrc_collision:
+   * @rtpbin: the object which received the signal
+   * @session: the session
+   * @ssrc: the SSRC 
+   *
+   * Notify when we have an SSRC collision
+   */
+  gst_rtp_dec_signals[SIGNAL_ON_SSRC_COLLISION] =
+      g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_ssrc_collision),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+      G_TYPE_UINT);
+  /**
+   * GstRTPDec::on-ssrc_validated:
+   * @rtpbin: the object which received the signal
+   * @session: the session
+   * @ssrc: the SSRC 
+   *
+   * Notify of a new SSRC that became validated.
+   */
+  gst_rtp_dec_signals[SIGNAL_ON_SSRC_VALIDATED] =
+      g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_ssrc_validated),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+      G_TYPE_UINT);
+
+  /**
+   * GstRTPDec::on-bye-ssrc:
+   * @rtpbin: the object which received the signal
+   * @session: the session
+   * @ssrc: the SSRC 
+   *
+   * Notify of an SSRC that became inactive because of a BYE packet.
+   */
+  gst_rtp_dec_signals[SIGNAL_ON_BYE_SSRC] =
+      g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_bye_ssrc),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+      G_TYPE_UINT);
+  /**
+   * GstRTPDec::on-bye-timeout:
+   * @rtpbin: the object which received the signal
+   * @session: the session
+   * @ssrc: the SSRC 
+   *
+   * Notify of an SSRC that has timed out because of BYE
+   */
+  gst_rtp_dec_signals[SIGNAL_ON_BYE_TIMEOUT] =
+      g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_bye_timeout),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+      G_TYPE_UINT);
+  /**
+   * GstRTPDec::on-timeout:
+   * @rtpbin: the object which received the signal
+   * @session: the session
+   * @ssrc: the SSRC 
+   *
+   * Notify of an SSRC that has timed out
+   */
+  gst_rtp_dec_signals[SIGNAL_ON_TIMEOUT] =
+      g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPDecClass, on_timeout),
+      NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2, G_TYPE_UINT,
+      G_TYPE_UINT);
+
+  gstelement_class->provide_clock =
+      GST_DEBUG_FUNCPTR (gst_rtp_dec_provide_clock);
+  gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_dec_change_state);
+  gstelement_class->request_new_pad =
+      GST_DEBUG_FUNCPTR (gst_rtp_dec_request_new_pad);
+  gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_dec_release_pad);
+
+  /* sink pads */
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_dec_recv_rtp_sink_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_dec_recv_rtcp_sink_template));
+  /* src pads */
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_dec_recv_rtp_src_template));
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&gst_rtp_dec_rtcp_src_template));
+
+  gst_element_class_set_static_metadata (gstelement_class, "RTP Decoder",
+      "Codec/Parser/Network",
+      "Accepts raw RTP and RTCP packets and sends them forward",
+      "Wim Taymans <wim.taymans@gmail.com>");
+}
+
+static void
+gst_rtp_dec_init (GstRTPDec * rtpdec)
+{
+  rtpdec->provided_clock = gst_system_clock_obtain ();
+  rtpdec->latency = DEFAULT_LATENCY_MS;
+
+  GST_OBJECT_FLAG_SET (rtpdec, GST_ELEMENT_FLAG_PROVIDE_CLOCK);
+}
+
+static void
+gst_rtp_dec_finalize (GObject * object)
+{
+  GstRTPDec *rtpdec;
+
+  rtpdec = GST_RTP_DEC (object);
+
+  g_slist_foreach (rtpdec->sessions, (GFunc) free_session, NULL);
+  g_slist_free (rtpdec->sessions);
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static gboolean
+gst_rtp_dec_query_src (GstPad * pad, GstObject * parent, GstQuery * query)
+{
+  gboolean res;
+
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_LATENCY:
+    {
+      /* we pretend to be live with a 3 second latency */
+      /* FIXME: Do we really have infinite maximum latency? */
+      gst_query_set_latency (query, TRUE, 3 * GST_SECOND, -1);
+      res = TRUE;
+      break;
+    }
+    default:
+      res = gst_pad_query_default (pad, parent, query);
+      break;
+  }
+  return res;
+}
+
+static GstFlowReturn
+gst_rtp_dec_chain_rtp (GstPad * pad, GstObject * parent, GstBuffer * buffer)
+{
+  GstFlowReturn res;
+  GstRTPDec *rtpdec;
+  GstRTPDecSession *session;
+  guint32 ssrc;
+  guint8 pt;
+  GstRTPBuffer rtp = { NULL, };
+
+  rtpdec = GST_RTP_DEC (parent);
+
+  GST_DEBUG_OBJECT (rtpdec, "got rtp packet");
+
+  if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
+    goto bad_packet;
+
+  ssrc = gst_rtp_buffer_get_ssrc (&rtp);
+  pt = gst_rtp_buffer_get_payload_type (&rtp);
+  gst_rtp_buffer_unmap (&rtp);
+
+  GST_DEBUG_OBJECT (rtpdec, "SSRC %08x, PT %d", ssrc, pt);
+
+  /* find session */
+  session = gst_pad_get_element_private (pad);
+
+  /* see if we have the pad */
+  if (!session->active) {
+    GstPadTemplate *templ;
+    GstElementClass *klass;
+    gchar *name;
+    GstCaps *caps;
+    GValue ret = { 0 };
+    GValue args[3] = { {0}
+    , {0}
+    , {0}
+    };
+
+    GST_DEBUG_OBJECT (rtpdec, "creating stream");
+
+    session->ssrc = ssrc;
+    session->pt = pt;
+
+    /* get pt map */
+    g_value_init (&args[0], GST_TYPE_ELEMENT);
+    g_value_set_object (&args[0], rtpdec);
+    g_value_init (&args[1], G_TYPE_UINT);
+    g_value_set_uint (&args[1], session->id);
+    g_value_init (&args[2], G_TYPE_UINT);
+    g_value_set_uint (&args[2], pt);
+
+    g_value_init (&ret, GST_TYPE_CAPS);
+    g_value_set_boxed (&ret, NULL);
+
+    g_signal_emitv (args, gst_rtp_dec_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret);
+
+    caps = (GstCaps *) g_value_get_boxed (&ret);
+
+    name = g_strdup_printf ("recv_rtp_src_%u_%u_%u", session->id, ssrc, pt);
+    klass = GST_ELEMENT_GET_CLASS (rtpdec);
+    templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
+    session->recv_rtp_src = gst_pad_new_from_template (templ, name);
+    g_free (name);
+
+    gst_pad_set_caps (session->recv_rtp_src, caps);
+
+    gst_pad_set_element_private (session->recv_rtp_src, session);
+    gst_pad_set_query_function (session->recv_rtp_src, gst_rtp_dec_query_src);
+    gst_pad_set_active (session->recv_rtp_src, TRUE);
+    gst_element_add_pad (GST_ELEMENT_CAST (rtpdec), session->recv_rtp_src);
+
+    session->active = TRUE;
+  }
+
+  res = gst_pad_push (session->recv_rtp_src, buffer);
+
+  return res;
+
+bad_packet:
+  {
+    GST_ELEMENT_WARNING (rtpdec, STREAM, DECODE, (NULL),
+        ("RTP packet did not validate, dropping"));
+    gst_buffer_unref (buffer);
+    return GST_FLOW_OK;
+  }
+}
+
+static GstFlowReturn
+gst_rtp_dec_chain_rtcp (GstPad * pad, GstObject * parent, GstBuffer * buffer)
+{
+  GstRTPDec *src;
+
+#ifdef HAVE_RTCP
+  gboolean valid;
+  GstRTCPPacket packet;
+  gboolean more;
+#endif
+
+  src = GST_RTP_DEC (parent);
+
+  GST_DEBUG_OBJECT (src, "got rtcp packet");
+
+#ifdef HAVE_RTCP
+  valid = gst_rtcp_buffer_validate (buffer);
+  if (!valid)
+    goto bad_packet;
+
+  /* position on first packet */
+  more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
+  while (more) {
+    switch (gst_rtcp_packet_get_type (&packet)) {
+      case GST_RTCP_TYPE_SR:
+      {
+        guint32 ssrc, rtptime, packet_count, octet_count;
+        guint64 ntptime;
+        guint count, i;
+
+        gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, &rtptime,
+            &packet_count, &octet_count);
+
+        GST_DEBUG_OBJECT (src,
+            "got SR packet: SSRC %08x, NTP %" G_GUINT64_FORMAT
+            ", RTP %u, PC %u, OC %u", ssrc, ntptime, rtptime, packet_count,
+            octet_count);
+
+        count = gst_rtcp_packet_get_rb_count (&packet);
+        for (i = 0; i < count; i++) {
+          guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
+          guint8 fractionlost;
+          gint32 packetslost;
+
+          gst_rtcp_packet_get_rb (&packet, i, &ssrc, &fractionlost,
+              &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
+
+          GST_DEBUG_OBJECT (src, "got RB packet %d: SSRC %08x, FL %u"
+              ", PL %u, HS %u, JITTER %u, LSR %u, DLSR %u", ssrc, fractionlost,
+              packetslost, exthighestseq, jitter, lsr, dlsr);
+        }
+        break;
+      }
+      case GST_RTCP_TYPE_RR:
+      {
+        guint32 ssrc;
+        guint count, i;
+
+        ssrc = gst_rtcp_packet_rr_get_ssrc (&packet);
+
+        GST_DEBUG_OBJECT (src, "got RR packet: SSRC %08x", ssrc);
+
+        count = gst_rtcp_packet_get_rb_count (&packet);
+        for (i = 0; i < count; i++) {
+          guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
+          guint8 fractionlost;
+          gint32 packetslost;
+
+          gst_rtcp_packet_get_rb (&packet, i, &ssrc, &fractionlost,
+              &packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
+
+          GST_DEBUG_OBJECT (src, "got RB packet %d: SSRC %08x, FL %u"
+              ", PL %u, HS %u, JITTER %u, LSR %u, DLSR %u", ssrc, fractionlost,
+              packetslost, exthighestseq, jitter, lsr, dlsr);
+        }
+        break;
+      }
+      case GST_RTCP_TYPE_SDES:
+      {
+        guint chunks, i, j;
+        gboolean more_chunks, more_items;
+
+        chunks = gst_rtcp_packet_sdes_get_chunk_count (&packet);
+        GST_DEBUG_OBJECT (src, "got SDES packet with %d chunks", chunks);
+
+        more_chunks = gst_rtcp_packet_sdes_first_chunk (&packet);
+        i = 0;
+        while (more_chunks) {
+          guint32 ssrc;
+
+          ssrc = gst_rtcp_packet_sdes_get_ssrc (&packet);
+
+          GST_DEBUG_OBJECT (src, "chunk %d, SSRC %08x", i, ssrc);
+
+          more_items = gst_rtcp_packet_sdes_first_item (&packet);
+          j = 0;
+          while (more_items) {
+            GstRTCPSDESType type;
+            guint8 len;
+            gchar *data;
+
+            gst_rtcp_packet_sdes_get_item (&packet, &type, &len, &data);
+
+            GST_DEBUG_OBJECT (src, "item %d, type %d, len %d, data %s", j,
+                type, len, data);
+
+            more_items = gst_rtcp_packet_sdes_next_item (&packet);
+            j++;
+          }
+          more_chunks = gst_rtcp_packet_sdes_next_chunk (&packet);
+          i++;
+        }
+        break;
+      }
+      case GST_RTCP_TYPE_BYE:
+      {
+        guint count, i;
+        gchar *reason;
+
+        reason = gst_rtcp_packet_bye_get_reason (&packet);
+        GST_DEBUG_OBJECT (src, "got BYE packet (reason: %s)",
+            GST_STR_NULL (reason));
+        g_free (reason);
+
+        count = gst_rtcp_packet_bye_get_ssrc_count (&packet);
+        for (i = 0; i < count; i++) {
+          guint32 ssrc;
+
+
+          ssrc = gst_rtcp_packet_bye_get_nth_ssrc (&packet, i);
+
+          GST_DEBUG_OBJECT (src, "SSRC: %08x", ssrc);
+        }
+        break;
+      }
+      case GST_RTCP_TYPE_APP:
+        GST_DEBUG_OBJECT (src, "got APP packet");
+        break;
+      default:
+        GST_WARNING_OBJECT (src, "got unknown RTCP packet");
+        break;
+    }
+    more = gst_rtcp_packet_move_to_next (&packet);
+  }
+  gst_buffer_unref (buffer);
+  return GST_FLOW_OK;
+
+bad_packet:
+  {
+    GST_WARNING_OBJECT (src, "got invalid RTCP packet");
+    gst_buffer_unref (buffer);
+    return GST_FLOW_OK;
+  }
+#else
+  gst_buffer_unref (buffer);
+  return GST_FLOW_OK;
+#endif
+}
+
+static void
+gst_rtp_dec_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRTPDec *src;
+
+  src = GST_RTP_DEC (object);
+
+  switch (prop_id) {
+    case PROP_LATENCY:
+      src->latency = g_value_get_uint (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rtp_dec_get_property (GObject * object, guint prop_id, GValue * value,
+    GParamSpec * pspec)
+{
+  GstRTPDec *src;
+
+  src = GST_RTP_DEC (object);
+
+  switch (prop_id) {
+    case PROP_LATENCY:
+      g_value_set_uint (value, src->latency);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstClock *
+gst_rtp_dec_provide_clock (GstElement * element)
+{
+  GstRTPDec *rtpdec;
+
+  rtpdec = GST_RTP_DEC (element);
+
+  return GST_CLOCK_CAST (gst_object_ref (rtpdec->provided_clock));
+}
+
+static GstStateChangeReturn
+gst_rtp_dec_change_state (GstElement * element, GstStateChange transition)
+{
+  GstStateChangeReturn ret;
+
+  switch (transition) {
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      /* we're NO_PREROLL when going to PAUSED */
+      ret = GST_STATE_CHANGE_NO_PREROLL;
+      break;
+    default:
+      break;
+  }
+
+  return ret;
+}
+
+/* Create a pad for receiving RTP for the session in @name
+ */
+static GstPad *
+create_recv_rtp (GstRTPDec * rtpdec, GstPadTemplate * templ, const gchar * name)
+{
+  guint sessid;
+  GstRTPDecSession *session;
+
+  /* first get the session number */
+  if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
+    goto no_name;
+
+  GST_DEBUG_OBJECT (rtpdec, "finding session %d", sessid);
+
+  /* get or create session */
+  session = find_session_by_id (rtpdec, sessid);
+  if (!session) {
+    GST_DEBUG_OBJECT (rtpdec, "creating session %d", sessid);
+    /* create session now */
+    session = create_session (rtpdec, sessid);
+    if (session == NULL)
+      goto create_error;
+  }
+  /* check if pad was requested */
+  if (session->recv_rtp_sink != NULL)
+    goto existed;
+
+  GST_DEBUG_OBJECT (rtpdec, "getting RTP sink pad");
+
+  session->recv_rtp_sink = gst_pad_new_from_template (templ, name);
+  gst_pad_set_element_private (session->recv_rtp_sink, session);
+  gst_pad_set_chain_function (session->recv_rtp_sink, gst_rtp_dec_chain_rtp);
+  gst_pad_set_active (session->recv_rtp_sink, TRUE);
+  gst_element_add_pad (GST_ELEMENT_CAST (rtpdec), session->recv_rtp_sink);
+
+  return session->recv_rtp_sink;
+
+  /* ERRORS */
+no_name:
+  {
+    g_warning ("rtpdec: invalid name given");
+    return NULL;
+  }
+create_error:
+  {
+    /* create_session already warned */
+    return NULL;
+  }
+existed:
+  {
+    g_warning ("rtpdec: recv_rtp pad already requested for session %d", sessid);
+    return NULL;
+  }
+}
+
+/* Create a pad for receiving RTCP for the session in @name
+ */
+static GstPad *
+create_recv_rtcp (GstRTPDec * rtpdec, GstPadTemplate * templ,
+    const gchar * name)
+{
+  guint sessid;
+  GstRTPDecSession *session;
+
+  /* first get the session number */
+  if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
+    goto no_name;
+
+  GST_DEBUG_OBJECT (rtpdec, "finding session %d", sessid);
+
+  /* get the session, it must exist or we error */
+  session = find_session_by_id (rtpdec, sessid);
+  if (!session)
+    goto no_session;
+
+  /* check if pad was requested */
+  if (session->recv_rtcp_sink != NULL)
+    goto existed;
+
+  GST_DEBUG_OBJECT (rtpdec, "getting RTCP sink pad");
+
+  session->recv_rtcp_sink = gst_pad_new_from_template (templ, name);
+  gst_pad_set_element_private (session->recv_rtp_sink, session);
+  gst_pad_set_chain_function (session->recv_rtcp_sink, gst_rtp_dec_chain_rtcp);
+  gst_pad_set_active (session->recv_rtcp_sink, TRUE);
+  gst_element_add_pad (GST_ELEMENT_CAST (rtpdec), session->recv_rtcp_sink);
+
+  return session->recv_rtcp_sink;
+
+  /* ERRORS */
+no_name:
+  {
+    g_warning ("rtpdec: invalid name given");
+    return NULL;
+  }
+no_session:
+  {
+    g_warning ("rtpdec: no session with id %d", sessid);
+    return NULL;
+  }
+existed:
+  {
+    g_warning ("rtpdec: recv_rtcp pad already requested for session %d",
+        sessid);
+    return NULL;
+  }
+}
+
+/* Create a pad for sending RTCP for the session in @name
+ */
+static GstPad *
+create_rtcp (GstRTPDec * rtpdec, GstPadTemplate * templ, const gchar * name)
+{
+  guint sessid;
+  GstRTPDecSession *session;
+
+  /* first get the session number */
+  if (name == NULL || sscanf (name, "rtcp_src_%u", &sessid) != 1)
+    goto no_name;
+
+  /* get or create session */
+  session = find_session_by_id (rtpdec, sessid);
+  if (!session)
+    goto no_session;
+
+  /* check if pad was requested */
+  if (session->rtcp_src != NULL)
+    goto existed;
+
+  session->rtcp_src = gst_pad_new_from_template (templ, name);
+  gst_pad_set_active (session->rtcp_src, TRUE);
+  gst_element_add_pad (GST_ELEMENT_CAST (rtpdec), session->rtcp_src);
+
+  return session->rtcp_src;
+
+  /* ERRORS */
+no_name:
+  {
+    g_warning ("rtpdec: invalid name given");
+    return NULL;
+  }
+no_session:
+  {
+    g_warning ("rtpdec: session with id %d does not exist", sessid);
+    return NULL;
+  }
+existed:
+  {
+    g_warning ("rtpdec: rtcp_src pad already requested for session %d", sessid);
+    return NULL;
+  }
+}
+
+/* 
+ */
+static GstPad *
+gst_rtp_dec_request_new_pad (GstElement * element,
+    GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
+{
+  GstRTPDec *rtpdec;
+  GstElementClass *klass;
+  GstPad *result;
+
+  g_return_val_if_fail (templ != NULL, NULL);
+  g_return_val_if_fail (GST_IS_RTP_DEC (element), NULL);
+
+  rtpdec = GST_RTP_DEC (element);
+  klass = GST_ELEMENT_GET_CLASS (element);
+
+  /* figure out the template */
+  if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u")) {
+    result = create_recv_rtp (rtpdec, templ, name);
+  } else if (templ == gst_element_class_get_pad_template (klass,
+          "recv_rtcp_sink_%u")) {
+    result = create_recv_rtcp (rtpdec, templ, name);
+  } else if (templ == gst_element_class_get_pad_template (klass, "rtcp_src_%u")) {
+    result = create_rtcp (rtpdec, templ, name);
+  } else
+    goto wrong_template;
+
+  return result;
+
+  /* ERRORS */
+wrong_template:
+  {
+    g_warning ("rtpdec: this is not our template");
+    return NULL;
+  }
+}
+
+static void
+gst_rtp_dec_release_pad (GstElement * element, GstPad * pad)
+{
+}
diff --git a/gst/rtsp/gstrtpdec.h b/gst/rtsp/gstrtpdec.h
new file mode 100755
index 0000000..5e83e23
--- /dev/null
+++ b/gst/rtsp/gstrtpdec.h
@@ -0,0 +1,88 @@
+/* GStreamer
+ * Copyright (C) <2005,2006> Wim Taymans <wim@fluendo.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_RTP_DEC_H__
+#define __GST_RTP_DEC_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTP_DEC  		(gst_rtp_dec_get_type())
+#define GST_IS_RTP_DEC(obj)  		(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_DEC))
+#define GST_IS_RTP_DEC_CLASS(klass) 	(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_DEC))
+#define GST_RTP_DEC(obj)  		(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_DEC, GstRTPDec))
+#define GST_RTP_DEC_CLASS(klass)  	(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_DEC, GstRTPDecClass))
+
+typedef struct _GstRTPDec GstRTPDec;
+typedef struct _GstRTPDecClass GstRTPDecClass;
+typedef struct _GstRTPDecSession GstRTPDecSession;
+
+struct _GstRTPDec {
+  GstElement  element;
+
+  guint       latency;
+  GSList     *sessions;
+  GstClock   *provided_clock;
+};
+
+struct _GstRTPDecClass {
+  GstElementClass parent_class;
+
+  /* get the caps for pt */
+  GstCaps* (*request_pt_map)  (GstRTPDec *rtpdec, guint session, guint pt);
+
+  void     (*clear_pt_map)    (GstRTPDec *rtpdec);
+
+  void     (*on_new_ssrc)       (GstRTPDec *rtpdec, guint session, guint32 ssrc);
+  void     (*on_ssrc_collision) (GstRTPDec *rtpdec, guint session, guint32 ssrc);
+  void     (*on_ssrc_validated) (GstRTPDec *rtpdec, guint session, guint32 ssrc);
+  void     (*on_bye_ssrc)       (GstRTPDec *rtpdec, guint session, guint32 ssrc);
+  void     (*on_bye_timeout)    (GstRTPDec *rtpdec, guint session, guint32 ssrc);
+  void     (*on_timeout)        (GstRTPDec *rtpdec, guint session, guint32 ssrc);
+};
+
+GType gst_rtp_dec_get_type(void);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_DEC_H__ */
diff --git a/gst/rtsp/gstrtsp.c b/gst/rtsp/gstrtsp.c
new file mode 100755
index 0000000..a2d568a
--- /dev/null
+++ b/gst/rtsp/gstrtsp.c
@@ -0,0 +1,74 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *               <2006> Wim Taymans <wim@fluendo.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gst/gst-i18n-plugin.h"
+
+#include "gstrtpdec.h"
+#include "gstrtspsrc.h"
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+#ifdef ENABLE_NLS
+  bindtextdomain (GETTEXT_PACKAGE, LOCALEDIR);
+  bind_textdomain_codeset (GETTEXT_PACKAGE, "UTF-8");
+#endif /* ENABLE_NLS */
+
+  if (!gst_element_register (plugin, "rtspsrc", GST_RANK_PRIMARY,
+          GST_TYPE_RTSPSRC))
+    return FALSE;
+  if (!gst_element_register (plugin, "rtpdec", GST_RANK_NONE, GST_TYPE_RTP_DEC))
+    return FALSE;
+
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    rtsp,
+    "transfer data via RTSP",
+    plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
diff --git a/gst/rtsp/gstrtsp.h b/gst/rtsp/gstrtsp.h
new file mode 100755
index 0000000..e0f5ef8
--- /dev/null
+++ b/gst/rtsp/gstrtsp.h
@@ -0,0 +1,53 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *               <2006> Wim Taymans <wim@fluendo.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_RTSP_H__
+#define __GST_RTSP_H__
+
+G_BEGIN_DECLS
+
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_H__ */
+
diff --git a/gst/rtsp/gstrtspext.c b/gst/rtsp/gstrtspext.c
new file mode 100755
index 0000000..07b5a97
--- /dev/null
+++ b/gst/rtsp/gstrtspext.c
@@ -0,0 +1,268 @@
+/* GStreamer
+ * Copyright (C) <2006> Wim Taymans <wim@fluendo.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#include "gstrtspext.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtspext_debug);
+#define GST_CAT_DEFAULT (rtspext_debug)
+
+static GList *extensions;
+
+static gboolean
+gst_rtsp_ext_list_filter (GstPluginFeature * feature, gpointer user_data)
+{
+  GstElementFactory *factory;
+  guint rank;
+
+  /* we only care about element factories */
+  if (!GST_IS_ELEMENT_FACTORY (feature))
+    return FALSE;
+
+  factory = GST_ELEMENT_FACTORY (feature);
+
+  if (!gst_element_factory_has_interface (factory, "GstRTSPExtension"))
+    return FALSE;
+
+  /* only select elements with autoplugging rank */
+  rank = gst_plugin_feature_get_rank (feature);
+  if (rank < GST_RANK_MARGINAL)
+    return FALSE;
+
+  return TRUE;
+}
+
+void
+gst_rtsp_ext_list_init (void)
+{
+  GST_DEBUG_CATEGORY_INIT (rtspext_debug, "rtspext", 0, "RTSP extension");
+
+  /* get a list of all extensions */
+  extensions = gst_registry_feature_filter (gst_registry_get (),
+      (GstPluginFeatureFilter) gst_rtsp_ext_list_filter, FALSE, NULL);
+}
+
+GstRTSPExtensionList *
+gst_rtsp_ext_list_get (void)
+{
+  GstRTSPExtensionList *result;
+  GList *walk;
+
+  result = g_new0 (GstRTSPExtensionList, 1);
+
+  for (walk = extensions; walk; walk = g_list_next (walk)) {
+    GstElementFactory *factory = GST_ELEMENT_FACTORY (walk->data);
+    GstElement *element;
+
+    element = gst_element_factory_create (factory, NULL);
+    if (!element) {
+      GST_ERROR ("could not create extension instance");
+      continue;
+    }
+
+    GST_DEBUG ("added extension interface for '%s'",
+        GST_ELEMENT_NAME (element));
+    result->extensions = g_list_prepend (result->extensions, element);
+  }
+  return result;
+}
+
+void
+gst_rtsp_ext_list_free (GstRTSPExtensionList * ext)
+{
+  GList *walk;
+
+  for (walk = ext->extensions; walk; walk = g_list_next (walk)) {
+    GstRTSPExtension *elem = (GstRTSPExtension *) walk->data;
+
+    gst_object_unref (GST_OBJECT_CAST (elem));
+  }
+  g_list_free (ext->extensions);
+  g_free (ext);
+}
+
+gboolean
+gst_rtsp_ext_list_detect_server (GstRTSPExtensionList * ext,
+    GstRTSPMessage * resp)
+{
+  GList *walk;
+  gboolean res = TRUE;
+
+  for (walk = ext->extensions; walk; walk = g_list_next (walk)) {
+    GstRTSPExtension *elem = (GstRTSPExtension *) walk->data;
+
+    res = gst_rtsp_extension_detect_server (elem, resp);
+  }
+  return res;
+}
+
+GstRTSPResult
+gst_rtsp_ext_list_before_send (GstRTSPExtensionList * ext, GstRTSPMessage * req)
+{
+  GList *walk;
+  GstRTSPResult res = GST_RTSP_OK;
+
+  for (walk = ext->extensions; walk; walk = g_list_next (walk)) {
+    GstRTSPExtension *elem = (GstRTSPExtension *) walk->data;
+
+    res = gst_rtsp_extension_before_send (elem, req);
+  }
+  return res;
+}
+
+GstRTSPResult
+gst_rtsp_ext_list_after_send (GstRTSPExtensionList * ext, GstRTSPMessage * req,
+    GstRTSPMessage * resp)
+{
+  GList *walk;
+  GstRTSPResult res = GST_RTSP_OK;
+
+  for (walk = ext->extensions; walk; walk = g_list_next (walk)) {
+    GstRTSPExtension *elem = (GstRTSPExtension *) walk->data;
+
+    res = gst_rtsp_extension_after_send (elem, req, resp);
+  }
+  return res;
+}
+
+GstRTSPResult
+gst_rtsp_ext_list_parse_sdp (GstRTSPExtensionList * ext, GstSDPMessage * sdp,
+    GstStructure * s)
+{
+  GList *walk;
+  GstRTSPResult res = GST_RTSP_OK;
+
+  for (walk = ext->extensions; walk; walk = g_list_next (walk)) {
+    GstRTSPExtension *elem = (GstRTSPExtension *) walk->data;
+
+    res = gst_rtsp_extension_parse_sdp (elem, sdp, s);
+  }
+  return res;
+}
+
+GstRTSPResult
+gst_rtsp_ext_list_setup_media (GstRTSPExtensionList * ext, GstSDPMedia * media)
+{
+  GList *walk;
+  GstRTSPResult res = GST_RTSP_OK;
+
+  for (walk = ext->extensions; walk; walk = g_list_next (walk)) {
+    GstRTSPExtension *elem = (GstRTSPExtension *) walk->data;
+
+    res = gst_rtsp_extension_setup_media (elem, media);
+  }
+  return res;
+}
+
+gboolean
+gst_rtsp_ext_list_configure_stream (GstRTSPExtensionList * ext, GstCaps * caps)
+{
+  GList *walk;
+  gboolean res = TRUE;
+
+  for (walk = ext->extensions; walk; walk = g_list_next (walk)) {
+    GstRTSPExtension *elem = (GstRTSPExtension *) walk->data;
+
+    res = gst_rtsp_extension_configure_stream (elem, caps);
+    if (!res)
+      break;
+  }
+  return res;
+}
+
+GstRTSPResult
+gst_rtsp_ext_list_get_transports (GstRTSPExtensionList * ext,
+    GstRTSPLowerTrans protocols, gchar ** transport)
+{
+  GList *walk;
+  GstRTSPResult res = GST_RTSP_OK;
+
+  for (walk = ext->extensions; walk; walk = g_list_next (walk)) {
+    GstRTSPExtension *elem = (GstRTSPExtension *) walk->data;
+
+    res = gst_rtsp_extension_get_transports (elem, protocols, transport);
+  }
+  return res;
+}
+
+GstRTSPResult
+gst_rtsp_ext_list_stream_select (GstRTSPExtensionList * ext, GstRTSPUrl * url)
+{
+  GList *walk;
+  GstRTSPResult res = GST_RTSP_OK;
+
+  for (walk = ext->extensions; walk; walk = g_list_next (walk)) {
+    GstRTSPExtension *elem = (GstRTSPExtension *) walk->data;
+
+    res = gst_rtsp_extension_stream_select (elem, url);
+  }
+  return res;
+}
+
+void
+gst_rtsp_ext_list_connect (GstRTSPExtensionList * ext,
+    const gchar * detailed_signal, GCallback c_handler, gpointer data)
+{
+  GList *walk;
+
+  for (walk = ext->extensions; walk; walk = g_list_next (walk)) {
+    GstRTSPExtension *elem = (GstRTSPExtension *) walk->data;
+
+    g_signal_connect (elem, detailed_signal, c_handler, data);
+  }
+}
+
+GstRTSPResult
+gst_rtsp_ext_list_receive_request (GstRTSPExtensionList * ext,
+    GstRTSPMessage * req)
+{
+  GList *walk;
+  GstRTSPResult res = GST_RTSP_ENOTIMPL;
+
+  for (walk = ext->extensions; walk; walk = g_list_next (walk)) {
+    GstRTSPExtension *elem = (GstRTSPExtension *) walk->data;
+
+    res = gst_rtsp_extension_receive_request (elem, req);
+    if (res != GST_RTSP_ENOTIMPL)
+      break;
+  }
+  return res;
+}
diff --git a/gst/rtsp/gstrtspext.h b/gst/rtsp/gstrtspext.h
new file mode 100755
index 0000000..2e87796
--- /dev/null
+++ b/gst/rtsp/gstrtspext.h
@@ -0,0 +1,83 @@
+/* GStreamer
+ * Copyright (C) <2006> Wim Taymans <wim@fluendo.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_RTSP_EXT_H__
+#define __GST_RTSP_EXT_H__
+
+#include <gst/gst.h>
+#include <gst/rtsp/gstrtspextension.h>
+
+G_BEGIN_DECLS
+
+typedef struct _GstRTSPExtensionList GstRTSPExtensionList;
+
+struct _GstRTSPExtensionList
+{
+  GList *extensions;
+};
+
+void                    gst_rtsp_ext_list_init    (void);
+
+GstRTSPExtensionList *  gst_rtsp_ext_list_get     (void);
+void                    gst_rtsp_ext_list_free    (GstRTSPExtensionList *ext);
+
+gboolean      gst_rtsp_ext_list_detect_server     (GstRTSPExtensionList *ext, GstRTSPMessage *resp);
+  
+GstRTSPResult gst_rtsp_ext_list_before_send       (GstRTSPExtensionList *ext, GstRTSPMessage *req);
+GstRTSPResult gst_rtsp_ext_list_after_send        (GstRTSPExtensionList *ext, GstRTSPMessage *req,
+                                                   GstRTSPMessage *resp);
+GstRTSPResult gst_rtsp_ext_list_parse_sdp         (GstRTSPExtensionList *ext, GstSDPMessage *sdp,
+                                                   GstStructure *s);
+GstRTSPResult gst_rtsp_ext_list_setup_media       (GstRTSPExtensionList *ext, GstSDPMedia *media);
+gboolean      gst_rtsp_ext_list_configure_stream  (GstRTSPExtensionList *ext, GstCaps *caps);
+GstRTSPResult gst_rtsp_ext_list_get_transports    (GstRTSPExtensionList *ext, GstRTSPLowerTrans protocols,
+                                                   gchar **transport);
+GstRTSPResult gst_rtsp_ext_list_stream_select     (GstRTSPExtensionList *ext, GstRTSPUrl *url);
+
+void          gst_rtsp_ext_list_connect           (GstRTSPExtensionList *ext,
+			                           const gchar *detailed_signal, GCallback c_handler,
+                                                   gpointer data);
+GstRTSPResult gst_rtsp_ext_list_receive_request   (GstRTSPExtensionList *ext, GstRTSPMessage *req);
+
+G_END_DECLS
+
+#endif /* __GST_RTSP_EXT_H__ */
diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c
new file mode 100755
index 0000000..b3b2a2e
--- /dev/null
+++ b/gst/rtsp/gstrtspsrc.c
@@ -0,0 +1,8317 @@
+/* GStreamer
+ * Copyright (C) <2005,2006> Wim Taymans <wim at fluendo dot com>
+ *               <2006> Lutz Mueller <lutz at topfrose dot de>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+/**
+ * SECTION:element-rtspsrc
+ *
+ * Makes a connection to an RTSP server and read the data.
+ * rtspsrc strictly follows RFC 2326 and therefore does not (yet) support
+ * RealMedia/Quicktime/Microsoft extensions.
+ *
+ * RTSP supports transport over TCP or UDP in unicast or multicast mode. By
+ * default rtspsrc will negotiate a connection in the following order:
+ * UDP unicast/UDP multicast/TCP. The order cannot be changed but the allowed
+ * protocols can be controlled with the #GstRTSPSrc:protocols property.
+ *
+ * rtspsrc currently understands SDP as the format of the session description.
+ * For each stream listed in the SDP a new rtp_stream\%d pad will be created
+ * with caps derived from the SDP media description. This is a caps of mime type
+ * "application/x-rtp" that can be connected to any available RTP depayloader
+ * element.
+ *
+ * rtspsrc will internally instantiate an RTP session manager element
+ * that will handle the RTCP messages to and from the server, jitter removal,
+ * packet reordering along with providing a clock for the pipeline.
+ * This feature is implemented using the gstrtpbin element.
+ *
+ * rtspsrc acts like a live source and will therefore only generate data in the
+ * PLAYING state.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch-1.0 rtspsrc location=rtsp://some.server/url ! fakesink
+ * ]| Establish a connection to an RTSP server and send the raw RTP packets to a
+ * fakesink.
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#ifdef HAVE_UNISTD_H
+#include <unistd.h>
+#endif /* HAVE_UNISTD_H */
+#include <stdlib.h>
+#include <string.h>
+#include <stdio.h>
+#include <stdarg.h>
+
+#include <gst/net/gstnet.h>
+#include <gst/sdp/gstsdpmessage.h>
+#include <gst/sdp/gstmikey.h>
+#include <gst/rtp/gstrtppayloads.h>
+
+#include "gst/gst-i18n-plugin.h"
+
+#include "gstrtspsrc.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtspsrc_debug);
+#define GST_CAT_DEFAULT (rtspsrc_debug)
+
+static GstStaticPadTemplate rtptemplate = GST_STATIC_PAD_TEMPLATE ("stream_%u",
+    GST_PAD_SRC,
+    GST_PAD_SOMETIMES,
+    GST_STATIC_CAPS ("application/x-rtp; application/x-rdt"));
+
+/* templates used internally */
+static GstStaticPadTemplate anysrctemplate =
+GST_STATIC_PAD_TEMPLATE ("internalsrc_%u",
+    GST_PAD_SRC,
+    GST_PAD_SOMETIMES,
+    GST_STATIC_CAPS_ANY);
+
+static GstStaticPadTemplate anysinktemplate =
+GST_STATIC_PAD_TEMPLATE ("internalsink_%u",
+    GST_PAD_SINK,
+    GST_PAD_SOMETIMES,
+    GST_STATIC_CAPS_ANY);
+
+enum
+{
+  SIGNAL_HANDLE_REQUEST,
+  SIGNAL_ON_SDP,
+  SIGNAL_SELECT_STREAM,
+  SIGNAL_NEW_MANAGER,
+  SIGNAL_REQUEST_RTCP_KEY,
+  LAST_SIGNAL
+};
+
+enum _GstRtspSrcRtcpSyncMode
+{
+  RTCP_SYNC_ALWAYS,
+  RTCP_SYNC_INITIAL,
+  RTCP_SYNC_RTP
+};
+
+enum _GstRtspSrcBufferMode
+{
+  BUFFER_MODE_NONE,
+  BUFFER_MODE_SLAVE,
+  BUFFER_MODE_BUFFER,
+  BUFFER_MODE_AUTO,
+  BUFFER_MODE_SYNCED
+};
+
+#define GST_TYPE_RTSP_SRC_BUFFER_MODE (gst_rtsp_src_buffer_mode_get_type())
+static GType
+gst_rtsp_src_buffer_mode_get_type (void)
+{
+  static GType buffer_mode_type = 0;
+  static const GEnumValue buffer_modes[] = {
+    {BUFFER_MODE_NONE, "Only use RTP timestamps", "none"},
+    {BUFFER_MODE_SLAVE, "Slave receiver to sender clock", "slave"},
+    {BUFFER_MODE_BUFFER, "Do low/high watermark buffering", "buffer"},
+    {BUFFER_MODE_AUTO, "Choose mode depending on stream live", "auto"},
+    {BUFFER_MODE_SYNCED, "Synchronized sender and receiver clocks", "synced"},
+    {0, NULL, NULL},
+  };
+
+  if (!buffer_mode_type) {
+    buffer_mode_type =
+        g_enum_register_static ("GstRTSPSrcBufferMode", buffer_modes);
+  }
+  return buffer_mode_type;
+}
+
+#define AES_128_KEY_LEN 16
+#define AES_256_KEY_LEN 32
+
+#define HMAC_32_KEY_LEN 4
+#define HMAC_80_KEY_LEN 10
+
+#define DEFAULT_LOCATION         NULL
+#define DEFAULT_PROTOCOLS        GST_RTSP_LOWER_TRANS_UDP | GST_RTSP_LOWER_TRANS_UDP_MCAST | GST_RTSP_LOWER_TRANS_TCP
+#define DEFAULT_DEBUG            FALSE
+#define DEFAULT_RETRY            20
+#define DEFAULT_TIMEOUT          5000000
+#define DEFAULT_UDP_BUFFER_SIZE  0x80000
+#define DEFAULT_TCP_TIMEOUT      20000000
+#define DEFAULT_LATENCY_MS       2000
+#define DEFAULT_DROP_ON_LATENCY  FALSE
+#define DEFAULT_CONNECTION_SPEED 0
+#define DEFAULT_NAT_METHOD       GST_RTSP_NAT_DUMMY
+#define DEFAULT_DO_RTCP          TRUE
+#define DEFAULT_DO_RTSP_KEEP_ALIVE       TRUE
+#define DEFAULT_PROXY            NULL
+#define DEFAULT_RTP_BLOCKSIZE    0
+#define DEFAULT_USER_ID          NULL
+#define DEFAULT_USER_PW          NULL
+#define DEFAULT_BUFFER_MODE      BUFFER_MODE_AUTO
+#define DEFAULT_PORT_RANGE       NULL
+#define DEFAULT_SHORT_HEADER     FALSE
+#define DEFAULT_PROBATION        2
+#define DEFAULT_UDP_RECONNECT    TRUE
+#define DEFAULT_MULTICAST_IFACE  NULL
+#define DEFAULT_NTP_SYNC         FALSE
+#define DEFAULT_USE_PIPELINE_CLOCK       FALSE
+#define DEFAULT_TLS_VALIDATION_FLAGS     G_TLS_CERTIFICATE_VALIDATE_ALL
+#define DEFAULT_TLS_DATABASE     NULL
+#define DEFAULT_DO_RETRANSMISSION        TRUE
+
+#ifdef GST_EXT_RTSP_MODIFICATION
+#define DEFAULT_START_POSITION   0
+#endif
+
+enum
+{
+  PROP_0,
+  PROP_LOCATION,
+  PROP_PROTOCOLS,
+  PROP_DEBUG,
+  PROP_RETRY,
+  PROP_TIMEOUT,
+#ifdef GST_EXT_RTSP_MODIFICATION
+  PROP_START_POSITION,
+#endif
+  PROP_TCP_TIMEOUT,
+  PROP_LATENCY,
+  PROP_DROP_ON_LATENCY,
+  PROP_CONNECTION_SPEED,
+  PROP_NAT_METHOD,
+  PROP_DO_RTCP,
+  PROP_DO_RTSP_KEEP_ALIVE,
+  PROP_PROXY,
+  PROP_PROXY_ID,
+  PROP_PROXY_PW,
+  PROP_RTP_BLOCKSIZE,
+  PROP_USER_ID,
+  PROP_USER_PW,
+  PROP_BUFFER_MODE,
+  PROP_PORT_RANGE,
+  PROP_UDP_BUFFER_SIZE,
+  PROP_SHORT_HEADER,
+  PROP_PROBATION,
+  PROP_UDP_RECONNECT,
+  PROP_MULTICAST_IFACE,
+  PROP_NTP_SYNC,
+  PROP_USE_PIPELINE_CLOCK,
+  PROP_SDES,
+  PROP_TLS_VALIDATION_FLAGS,
+  PROP_TLS_DATABASE,
+  PROP_DO_RETRANSMISSION
+};
+
+#define GST_TYPE_RTSP_NAT_METHOD (gst_rtsp_nat_method_get_type())
+static GType
+gst_rtsp_nat_method_get_type (void)
+{
+  static GType rtsp_nat_method_type = 0;
+  static const GEnumValue rtsp_nat_method[] = {
+    {GST_RTSP_NAT_NONE, "None", "none"},
+    {GST_RTSP_NAT_DUMMY, "Send Dummy packets", "dummy"},
+    {0, NULL, NULL},
+  };
+
+  if (!rtsp_nat_method_type) {
+    rtsp_nat_method_type =
+        g_enum_register_static ("GstRTSPNatMethod", rtsp_nat_method);
+  }
+  return rtsp_nat_method_type;
+}
+
+static void gst_rtspsrc_finalize (GObject * object);
+
+static void gst_rtspsrc_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rtspsrc_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static GstClock *gst_rtspsrc_provide_clock (GstElement * element);
+
+static void gst_rtspsrc_uri_handler_init (gpointer g_iface,
+    gpointer iface_data);
+
+static void gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes,
+    GstCaps * caps);
+
+static gboolean gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy);
+static void gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout);
+
+static GstCaps *gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media);
+
+static GstStateChangeReturn gst_rtspsrc_change_state (GstElement * element,
+    GstStateChange transition);
+static gboolean gst_rtspsrc_send_event (GstElement * element, GstEvent * event);
+static void gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message);
+
+static gboolean gst_rtspsrc_setup_auth (GstRTSPSrc * src,
+    GstRTSPMessage * response);
+
+static gboolean gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd,
+    gint mask);
+static GstRTSPResult gst_rtspsrc_send_cb (GstRTSPExtension * ext,
+    GstRTSPMessage * request, GstRTSPMessage * response, GstRTSPSrc * src);
+
+static GstRTSPResult gst_rtspsrc_open (GstRTSPSrc * src, gboolean async);
+static GstRTSPResult gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment,
+    gboolean async);
+static GstRTSPResult gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async);
+static GstRTSPResult gst_rtspsrc_close (GstRTSPSrc * src, gboolean async,
+    gboolean only_close);
+
+static gboolean gst_rtspsrc_uri_set_uri (GstURIHandler * handler,
+    const gchar * uri, GError ** error);
+static gchar *gst_rtspsrc_uri_get_uri (GstURIHandler * handler);
+
+static gboolean gst_rtspsrc_activate_streams (GstRTSPSrc * src);
+static gboolean gst_rtspsrc_loop (GstRTSPSrc * src);
+static gboolean gst_rtspsrc_stream_push_event (GstRTSPSrc * src,
+    GstRTSPStream * stream, GstEvent * event);
+static gboolean gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event);
+static void gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush);
+
+typedef struct
+{
+  guint8 pt;
+  GstCaps *caps;
+} PtMapItem;
+
+/* commands we send to out loop to notify it of events */
+#define CMD_OPEN	(1 << 0)
+#define CMD_PLAY	(1 << 1)
+#define CMD_PAUSE	(1 << 2)
+#define CMD_CLOSE	(1 << 3)
+#define CMD_WAIT	(1 << 4)
+#define CMD_RECONNECT	(1 << 5)
+#define CMD_LOOP	(1 << 6)
+
+/* mask for all commands */
+#define CMD_ALL         ((CMD_LOOP << 1) - 1)
+
+#define GST_ELEMENT_PROGRESS(el, type, code, text)      \
+G_STMT_START {                                          \
+  gchar *__txt = _gst_element_error_printf text;        \
+  gst_element_post_message (GST_ELEMENT_CAST (el),      \
+      gst_message_new_progress (GST_OBJECT_CAST (el),   \
+          GST_PROGRESS_TYPE_ ##type, code, __txt));     \
+  g_free (__txt);                                       \
+} G_STMT_END
+
+static guint gst_rtspsrc_signals[LAST_SIGNAL] = { 0 };
+
+#define gst_rtspsrc_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstRTSPSrc, gst_rtspsrc, GST_TYPE_BIN,
+    G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtspsrc_uri_handler_init));
+
+#ifndef GST_DISABLE_GST_DEBUG
+static inline const char *
+cmd_to_string (guint cmd)
+{
+  switch (cmd) {
+    case CMD_OPEN:
+      return "OPEN";
+    case CMD_PLAY:
+      return "PLAY";
+    case CMD_PAUSE:
+      return "PAUSE";
+    case CMD_CLOSE:
+      return "CLOSE";
+    case CMD_WAIT:
+      return "WAIT";
+    case CMD_RECONNECT:
+      return "RECONNECT";
+    case CMD_LOOP:
+      return "LOOP";
+  }
+
+  return "unknown";
+}
+#endif
+
+static gboolean
+default_select_stream (GstRTSPSrc * src, guint id, GstCaps * caps)
+{
+  GST_DEBUG_OBJECT (src, "default handler");
+  return TRUE;
+}
+
+static gboolean
+select_stream_accum (GSignalInvocationHint * ihint,
+    GValue * return_accu, const GValue * handler_return, gpointer data)
+{
+  gboolean myboolean;
+
+  myboolean = g_value_get_boolean (handler_return);
+  GST_DEBUG ("accum %d", myboolean);
+  g_value_set_boolean (return_accu, myboolean);
+
+  /* stop emission if FALSE */
+  return myboolean;
+}
+
+static void
+gst_rtspsrc_class_init (GstRTSPSrcClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBinClass *gstbin_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbin_class = (GstBinClass *) klass;
+
+  GST_DEBUG_CATEGORY_INIT (rtspsrc_debug, "rtspsrc", 0, "RTSP src");
+
+  gobject_class->set_property = gst_rtspsrc_set_property;
+  gobject_class->get_property = gst_rtspsrc_get_property;
+
+  gobject_class->finalize = gst_rtspsrc_finalize;
+
+  g_object_class_install_property (gobject_class, PROP_LOCATION,
+      g_param_spec_string ("location", "RTSP Location",
+          "Location of the RTSP url to read",
+          DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_PROTOCOLS,
+      g_param_spec_flags ("protocols", "Protocols",
+          "Allowed lower transport protocols", GST_TYPE_RTSP_LOWER_TRANS,
+          DEFAULT_PROTOCOLS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_DEBUG,
+      g_param_spec_boolean ("debug", "Debug",
+          "Dump request and response messages to stdout",
+          DEFAULT_DEBUG, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_RETRY,
+      g_param_spec_uint ("retry", "Retry",
+          "Max number of retries when allocating RTP ports.",
+          0, G_MAXUINT16, DEFAULT_RETRY,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_TIMEOUT,
+      g_param_spec_uint64 ("timeout", "Timeout",
+          "Retry TCP transport after UDP timeout microseconds (0 = disabled)",
+          0, G_MAXUINT64, DEFAULT_TIMEOUT,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+#ifdef GST_EXT_RTSP_MODIFICATION
+  g_object_class_install_property (gobject_class, PROP_START_POSITION,
+      g_param_spec_uint64 ("pending-start-position", "set start position",
+          "Set start position before PLAYING request.",
+          0, G_MAXUINT64, DEFAULT_START_POSITION,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+#endif
+  g_object_class_install_property (gobject_class, PROP_TCP_TIMEOUT,
+      g_param_spec_uint64 ("tcp-timeout", "TCP Timeout",
+          "Fail after timeout microseconds on TCP connections (0 = disabled)",
+          0, G_MAXUINT64, DEFAULT_TCP_TIMEOUT,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_LATENCY,
+      g_param_spec_uint ("latency", "Buffer latency in ms",
+          "Amount of ms to buffer", 0, G_MAXUINT, DEFAULT_LATENCY_MS,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
+      g_param_spec_boolean ("drop-on-latency",
+          "Drop buffers when maximum latency is reached",
+          "Tells the jitterbuffer to never exceed the given latency in size",
+          DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_CONNECTION_SPEED,
+      g_param_spec_uint64 ("connection-speed", "Connection Speed",
+          "Network connection speed in kbps (0 = unknown)",
+          0, G_MAXUINT64 / 1000, DEFAULT_CONNECTION_SPEED,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_NAT_METHOD,
+      g_param_spec_enum ("nat-method", "NAT Method",
+          "Method to use for traversing firewalls and NAT",
+          GST_TYPE_RTSP_NAT_METHOD, DEFAULT_NAT_METHOD,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRTSPSrc:do-rtcp:
+   *
+   * Enable RTCP support. Some old server don't like RTCP and then this property
+   * needs to be set to FALSE.
+   */
+  g_object_class_install_property (gobject_class, PROP_DO_RTCP,
+      g_param_spec_boolean ("do-rtcp", "Do RTCP",
+          "Send RTCP packets, disable for old incompatible server.",
+          DEFAULT_DO_RTCP, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRTSPSrc:do-rtsp-keep-alive:
+   *
+   * Enable RTSP keep alive support. Some old server don't like RTSP
+   * keep alive and then this property needs to be set to FALSE.
+   */
+  g_object_class_install_property (gobject_class, PROP_DO_RTSP_KEEP_ALIVE,
+      g_param_spec_boolean ("do-rtsp-keep-alive", "Do RTSP Keep Alive",
+          "Send RTSP keep alive packets, disable for old incompatible server.",
+          DEFAULT_DO_RTSP_KEEP_ALIVE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRTSPSrc:proxy:
+   *
+   * Set the proxy parameters. This has to be a string of the format
+   * [http://][user:passwd@]host[:port].
+   */
+  g_object_class_install_property (gobject_class, PROP_PROXY,
+      g_param_spec_string ("proxy", "Proxy",
+          "Proxy settings for HTTP tunneling. Format: [http://][user:passwd@]host[:port]",
+          DEFAULT_PROXY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRTSPSrc:proxy-id:
+   *
+   * Sets the proxy URI user id for authentication. If the URI set via the
+   * "proxy" property contains a user-id already, that will take precedence.
+   *
+   * Since: 1.2
+   */
+  g_object_class_install_property (gobject_class, PROP_PROXY_ID,
+      g_param_spec_string ("proxy-id", "proxy-id",
+          "HTTP proxy URI user id for authentication", "",
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRTSPSrc:proxy-pw:
+   *
+   * Sets the proxy URI password for authentication. If the URI set via the
+   * "proxy" property contains a password already, that will take precedence.
+   *
+   * Since: 1.2
+   */
+  g_object_class_install_property (gobject_class, PROP_PROXY_PW,
+      g_param_spec_string ("proxy-pw", "proxy-pw",
+          "HTTP proxy URI user password for authentication", "",
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRTSPSrc:rtp-blocksize:
+   *
+   * RTP package size to suggest to server.
+   */
+  g_object_class_install_property (gobject_class, PROP_RTP_BLOCKSIZE,
+      g_param_spec_uint ("rtp-blocksize", "RTP Blocksize",
+          "RTP package size to suggest to server (0 = disabled)",
+          0, 65536, DEFAULT_RTP_BLOCKSIZE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class,
+      PROP_USER_ID,
+      g_param_spec_string ("user-id", "user-id",
+          "RTSP location URI user id for authentication", DEFAULT_USER_ID,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  g_object_class_install_property (gobject_class, PROP_USER_PW,
+      g_param_spec_string ("user-pw", "user-pw",
+          "RTSP location URI user password for authentication", DEFAULT_USER_PW,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRTSPSrc:buffer-mode:
+   *
+   * Control the buffering and timestamping mode used by the jitterbuffer.
+   */
+  g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
+      g_param_spec_enum ("buffer-mode", "Buffer Mode",
+          "Control the buffering algorithm in use",
+          GST_TYPE_RTSP_SRC_BUFFER_MODE, DEFAULT_BUFFER_MODE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRTSPSrc:port-range:
+   *
+   * Configure the client port numbers that can be used to recieve RTP and
+   * RTCP.
+   */
+  g_object_class_install_property (gobject_class, PROP_PORT_RANGE,
+      g_param_spec_string ("port-range", "Port range",
+          "Client port range that can be used to receive RTP and RTCP data, "
+          "eg. 3000-3005 (NULL = no restrictions)", DEFAULT_PORT_RANGE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRTSPSrc:udp-buffer-size:
+   *
+   * Size of the kernel UDP receive buffer in bytes.
+   */
+  g_object_class_install_property (gobject_class, PROP_UDP_BUFFER_SIZE,
+      g_param_spec_int ("udp-buffer-size", "UDP Buffer Size",
+          "Size of the kernel UDP receive buffer in bytes, 0=default",
+          0, G_MAXINT, DEFAULT_UDP_BUFFER_SIZE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRTSPSrc:short-header:
+   *
+   * Only send the basic RTSP headers for broken encoders.
+   */
+  g_object_class_install_property (gobject_class, PROP_SHORT_HEADER,
+      g_param_spec_boolean ("short-header", "Short Header",
+          "Only send the basic RTSP headers for broken encoders",
+          DEFAULT_SHORT_HEADER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_PROBATION,
+      g_param_spec_uint ("probation", "Number of probations",
+          "Consecutive packet sequence numbers to accept the source",
+          0, G_MAXUINT, DEFAULT_PROBATION,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_UDP_RECONNECT,
+      g_param_spec_boolean ("udp-reconnect", "Reconnect to the server",
+          "Reconnect to the server if RTSP connection is closed when doing UDP",
+          DEFAULT_UDP_RECONNECT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_MULTICAST_IFACE,
+      g_param_spec_string ("multicast-iface", "Multicast Interface",
+          "The network interface on which to join the multicast group",
+          DEFAULT_MULTICAST_IFACE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
+      g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
+          "Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
+      g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
+          "Use the pipeline running-time to set the NTP time in the RTCP SR messages",
+          DEFAULT_USE_PIPELINE_CLOCK,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_SDES,
+      g_param_spec_boxed ("sdes", "SDES",
+          "The SDES items of this session",
+          GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRTSPSrc::tls-validation-flags:
+   *
+   * TLS certificate validation flags used to validate server
+   * certificate.
+   *
+   * Since: 1.2.1
+   */
+  g_object_class_install_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
+      g_param_spec_flags ("tls-validation-flags", "TLS validation flags",
+          "TLS certificate validation flags used to validate the server certificate",
+          G_TYPE_TLS_CERTIFICATE_FLAGS, DEFAULT_TLS_VALIDATION_FLAGS,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRTSPSrc::tls-database:
+   *
+   * TLS database with anchor certificate authorities used to validate
+   * the server certificate.
+   *
+   * Since: 1.4
+   */
+  g_object_class_install_property (gobject_class, PROP_TLS_DATABASE,
+      g_param_spec_object ("tls-database", "TLS database",
+          "TLS database with anchor certificate authorities used to validate the server certificate",
+          G_TYPE_TLS_DATABASE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRTSPSrc::do-retransmission:
+   *
+   * Attempt to ask the server to retransmit lost packets according to RFC4588.
+   *
+   * Note: currently only works with SSRC-multiplexed retransmission streams
+   *
+   * Since: 1.6
+   */
+  g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
+      g_param_spec_boolean ("do-retransmission", "Retransmission",
+          "Ask the server to retransmit lost packets",
+          DEFAULT_DO_RETRANSMISSION,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstRTSPSrc::handle-request:
+   * @rtspsrc: a #GstRTSPSrc
+   * @request: a #GstRTSPMessage
+   * @response: a #GstRTSPMessage
+   *
+   * Handle a server request in @request and prepare @response.
+   *
+   * This signal is called from the streaming thread, you should therefore not
+   * do any state changes on @rtspsrc because this might deadlock. If you want
+   * to modify the state as a result of this signal, post a
+   * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
+   * in some other way.
+   *
+   * Since: 1.2
+   */
+  gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST] =
+      g_signal_new ("handle-request", G_TYPE_FROM_CLASS (klass), 0,
+      0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 2,
+      G_TYPE_POINTER, G_TYPE_POINTER);
+
+  /**
+   * GstRTSPSrc::on-sdp:
+   * @rtspsrc: a #GstRTSPSrc
+   * @sdp: a #GstSDPMessage
+   *
+   * Emited when the client has retrieved the SDP and before it configures the
+   * streams in the SDP. @sdp can be inspected and modified.
+   *
+   * This signal is called from the streaming thread, you should therefore not
+   * do any state changes on @rtspsrc because this might deadlock. If you want
+   * to modify the state as a result of this signal, post a
+   * #GST_MESSAGE_REQUEST_STATE message on the bus or signal the main thread
+   * in some other way.
+   *
+   * Since: 1.2
+   */
+  gst_rtspsrc_signals[SIGNAL_ON_SDP] =
+      g_signal_new ("on-sdp", G_TYPE_FROM_CLASS (klass), 0,
+      0, NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
+      GST_TYPE_SDP_MESSAGE | G_SIGNAL_TYPE_STATIC_SCOPE);
+
+  /**
+   * GstRTSPSrc::select-stream:
+   * @rtspsrc: a #GstRTSPSrc
+   * @num: the stream number
+   * @caps: the stream caps
+   *
+   * Emited before the client decides to configure the stream @num with
+   * @caps.
+   *
+   * Returns: %TRUE when the stream should be selected, %FALSE when the stream
+   * is to be ignored.
+   *
+   * Since: 1.2
+   */
+  gst_rtspsrc_signals[SIGNAL_SELECT_STREAM] =
+      g_signal_new_class_handler ("select-stream", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP,
+      (GCallback) default_select_stream, select_stream_accum, NULL,
+      g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2, G_TYPE_UINT,
+      GST_TYPE_CAPS);
+  /**
+   * GstRTSPSrc::new-manager:
+   * @rtspsrc: a #GstRTSPSrc
+   * @manager: a #GstElement
+   *
+   * Emited after a new manager (like rtpbin) was created and the default
+   * properties were configured.
+   *
+   * Since: 1.4
+   */
+  gst_rtspsrc_signals[SIGNAL_NEW_MANAGER] =
+      g_signal_new_class_handler ("new-manager", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_FIRST | G_SIGNAL_RUN_CLEANUP, 0, NULL, NULL,
+      g_cclosure_marshal_generic, G_TYPE_NONE, 1, GST_TYPE_ELEMENT);
+
+  /**
+   * GstRTSPSrc::request-rtcp-key:
+   * @rtspsrc: a #GstRTSPSrc
+   * @num: the stream number
+   *
+   * Signal emited to get the crypto parameters relevant to the RTCP
+   * stream. User should provide the key and the RTCP encryption ciphers
+   * and authentication, and return them wrapped in a GstCaps.
+   *
+   * Since: 1.4
+   */
+  gst_rtspsrc_signals[SIGNAL_REQUEST_RTCP_KEY] =
+      g_signal_new ("request-rtcp-key", G_TYPE_FROM_CLASS (klass),
+      G_SIGNAL_RUN_LAST, 0, NULL, NULL, NULL, GST_TYPE_CAPS, 1, G_TYPE_UINT);
+
+  gstelement_class->send_event = gst_rtspsrc_send_event;
+  gstelement_class->provide_clock = gst_rtspsrc_provide_clock;
+  gstelement_class->change_state = gst_rtspsrc_change_state;
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&rtptemplate));
+
+  gst_element_class_set_static_metadata (gstelement_class,
+      "RTSP packet receiver", "Source/Network",
+      "Receive data over the network via RTSP (RFC 2326)",
+      "Wim Taymans <wim@fluendo.com>, "
+      "Thijs Vermeir <thijs.vermeir@barco.com>, "
+      "Lutz Mueller <lutz@topfrose.de>");
+
+  gstbin_class->handle_message = gst_rtspsrc_handle_message;
+
+  gst_rtsp_ext_list_init ();
+}
+
+static void
+gst_rtspsrc_init (GstRTSPSrc * src)
+{
+  src->conninfo.location = g_strdup (DEFAULT_LOCATION);
+  src->protocols = DEFAULT_PROTOCOLS;
+  src->debug = DEFAULT_DEBUG;
+  src->retry = DEFAULT_RETRY;
+  src->udp_timeout = DEFAULT_TIMEOUT;
+#ifdef GST_EXT_RTSP_MODIFICATION
+  src->start_position = DEFAULT_START_POSITION;
+#endif
+  gst_rtspsrc_set_tcp_timeout (src, DEFAULT_TCP_TIMEOUT);
+  src->latency = DEFAULT_LATENCY_MS;
+  src->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
+  src->connection_speed = DEFAULT_CONNECTION_SPEED;
+  src->nat_method = DEFAULT_NAT_METHOD;
+  src->do_rtcp = DEFAULT_DO_RTCP;
+  src->do_rtsp_keep_alive = DEFAULT_DO_RTSP_KEEP_ALIVE;
+  gst_rtspsrc_set_proxy (src, DEFAULT_PROXY);
+  src->rtp_blocksize = DEFAULT_RTP_BLOCKSIZE;
+  src->user_id = g_strdup (DEFAULT_USER_ID);
+  src->user_pw = g_strdup (DEFAULT_USER_PW);
+  src->buffer_mode = DEFAULT_BUFFER_MODE;
+  src->client_port_range.min = 0;
+  src->client_port_range.max = 0;
+  src->udp_buffer_size = DEFAULT_UDP_BUFFER_SIZE;
+  src->short_header = DEFAULT_SHORT_HEADER;
+  src->probation = DEFAULT_PROBATION;
+  src->udp_reconnect = DEFAULT_UDP_RECONNECT;
+  src->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
+  src->ntp_sync = DEFAULT_NTP_SYNC;
+  src->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
+  src->sdes = NULL;
+  src->tls_validation_flags = DEFAULT_TLS_VALIDATION_FLAGS;
+  src->tls_database = DEFAULT_TLS_DATABASE;
+  src->do_retransmission = DEFAULT_DO_RETRANSMISSION;
+
+#ifdef GST_EXT_RTSP_MODIFICATION
+  g_mutex_init (&(src)->pause_lock);
+  g_cond_init (&(src)->open_end);
+#endif
+  /* get a list of all extensions */
+  src->extensions = gst_rtsp_ext_list_get ();
+
+  /* connect to send signal */
+  gst_rtsp_ext_list_connect (src->extensions, "send",
+      (GCallback) gst_rtspsrc_send_cb, src);
+
+  /* protects the streaming thread in interleaved mode or the polling
+   * thread in UDP mode. */
+  g_rec_mutex_init (&src->stream_rec_lock);
+
+  /* protects our state changes from multiple invocations */
+  g_rec_mutex_init (&src->state_rec_lock);
+
+  src->state = GST_RTSP_STATE_INVALID;
+
+  GST_OBJECT_FLAG_SET (src, GST_ELEMENT_FLAG_SOURCE);
+}
+
+static void
+gst_rtspsrc_finalize (GObject * object)
+{
+  GstRTSPSrc *rtspsrc;
+
+  rtspsrc = GST_RTSPSRC (object);
+
+  gst_rtsp_ext_list_free (rtspsrc->extensions);
+  g_free (rtspsrc->conninfo.location);
+  gst_rtsp_url_free (rtspsrc->conninfo.url);
+  g_free (rtspsrc->conninfo.url_str);
+  g_free (rtspsrc->user_id);
+  g_free (rtspsrc->user_pw);
+  g_free (rtspsrc->multi_iface);
+
+#ifdef GST_EXT_RTSP_MODIFICATION
+  g_mutex_clear (&(rtspsrc)->pause_lock);
+  g_cond_clear (&(rtspsrc)->open_end);
+#endif
+
+  if (rtspsrc->sdp) {
+    gst_sdp_message_free (rtspsrc->sdp);
+    rtspsrc->sdp = NULL;
+  }
+  if (rtspsrc->provided_clock)
+    gst_object_unref (rtspsrc->provided_clock);
+
+  if (rtspsrc->sdes)
+    gst_structure_free (rtspsrc->sdes);
+
+  if (rtspsrc->tls_database)
+    g_object_unref (rtspsrc->tls_database);
+
+  /* free locks */
+  g_rec_mutex_clear (&rtspsrc->stream_rec_lock);
+  g_rec_mutex_clear (&rtspsrc->state_rec_lock);
+
+  G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+static GstClock *
+gst_rtspsrc_provide_clock (GstElement * element)
+{
+  GstRTSPSrc *src = GST_RTSPSRC (element);
+  GstClock *clock;
+
+  if ((clock = src->provided_clock) != NULL)
+    gst_object_ref (clock);
+
+  return clock;
+}
+
+/* a proxy string of the format [user:passwd@]host[:port] */
+static gboolean
+gst_rtspsrc_set_proxy (GstRTSPSrc * rtsp, const gchar * proxy)
+{
+  gchar *p, *at, *col;
+
+  g_free (rtsp->proxy_user);
+  rtsp->proxy_user = NULL;
+  g_free (rtsp->proxy_passwd);
+  rtsp->proxy_passwd = NULL;
+  g_free (rtsp->proxy_host);
+  rtsp->proxy_host = NULL;
+  rtsp->proxy_port = 0;
+
+  p = (gchar *) proxy;
+
+  if (p == NULL)
+    return TRUE;
+
+  /* we allow http:// in front but ignore it */
+  if (g_str_has_prefix (p, "http://"))
+    p += 7;
+
+  at = strchr (p, '@');
+  if (at) {
+    /* look for user:passwd */
+    col = strchr (proxy, ':');
+    if (col == NULL || col > at)
+      return FALSE;
+
+    rtsp->proxy_user = g_strndup (p, col - p);
+    col++;
+    rtsp->proxy_passwd = g_strndup (col, at - col);
+
+    /* move to host */
+    p = at + 1;
+  } else {
+    if (rtsp->prop_proxy_id != NULL && *rtsp->prop_proxy_id != '\0')
+      rtsp->proxy_user = g_strdup (rtsp->prop_proxy_id);
+    if (rtsp->prop_proxy_pw != NULL && *rtsp->prop_proxy_pw != '\0')
+      rtsp->proxy_passwd = g_strdup (rtsp->prop_proxy_pw);
+    if (rtsp->proxy_user != NULL || rtsp->proxy_passwd != NULL) {
+      GST_LOG_OBJECT (rtsp, "set proxy user/pw from properties: %s:%s",
+          GST_STR_NULL (rtsp->proxy_user), GST_STR_NULL (rtsp->proxy_passwd));
+    }
+  }
+  col = strchr (p, ':');
+
+  if (col) {
+    /* everything before the colon is the hostname */
+    rtsp->proxy_host = g_strndup (p, col - p);
+    p = col + 1;
+    rtsp->proxy_port = strtoul (p, (char **) &p, 10);
+  } else {
+    rtsp->proxy_host = g_strdup (p);
+    rtsp->proxy_port = 8080;
+  }
+  return TRUE;
+}
+
+static void
+gst_rtspsrc_set_tcp_timeout (GstRTSPSrc * rtspsrc, guint64 timeout)
+{
+  rtspsrc->tcp_timeout.tv_sec = timeout / G_USEC_PER_SEC;
+  rtspsrc->tcp_timeout.tv_usec = timeout % G_USEC_PER_SEC;
+
+  if (timeout != 0)
+    rtspsrc->ptcp_timeout = &rtspsrc->tcp_timeout;
+  else
+    rtspsrc->ptcp_timeout = NULL;
+}
+
+static void
+gst_rtspsrc_set_property (GObject * object, guint prop_id, const GValue * value,
+    GParamSpec * pspec)
+{
+  GstRTSPSrc *rtspsrc;
+
+  rtspsrc = GST_RTSPSRC (object);
+
+  switch (prop_id) {
+    case PROP_LOCATION:
+      gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (rtspsrc),
+          g_value_get_string (value), NULL);
+      break;
+    case PROP_PROTOCOLS:
+      rtspsrc->protocols = g_value_get_flags (value);
+      break;
+    case PROP_DEBUG:
+      rtspsrc->debug = g_value_get_boolean (value);
+      break;
+    case PROP_RETRY:
+      rtspsrc->retry = g_value_get_uint (value);
+      break;
+    case PROP_TIMEOUT:
+      rtspsrc->udp_timeout = g_value_get_uint64 (value);
+      break;
+#ifdef GST_EXT_RTSP_MODIFICATION
+    case PROP_START_POSITION:
+      rtspsrc->start_position = g_value_get_uint64 (value);
+      break;
+#endif
+    case PROP_TCP_TIMEOUT:
+      gst_rtspsrc_set_tcp_timeout (rtspsrc, g_value_get_uint64 (value));
+      break;
+    case PROP_LATENCY:
+      rtspsrc->latency = g_value_get_uint (value);
+      break;
+    case PROP_DROP_ON_LATENCY:
+      rtspsrc->drop_on_latency = g_value_get_boolean (value);
+      break;
+    case PROP_CONNECTION_SPEED:
+      rtspsrc->connection_speed = g_value_get_uint64 (value);
+      break;
+    case PROP_NAT_METHOD:
+      rtspsrc->nat_method = g_value_get_enum (value);
+      break;
+    case PROP_DO_RTCP:
+      rtspsrc->do_rtcp = g_value_get_boolean (value);
+      break;
+    case PROP_DO_RTSP_KEEP_ALIVE:
+      rtspsrc->do_rtsp_keep_alive = g_value_get_boolean (value);
+      break;
+    case PROP_PROXY:
+      gst_rtspsrc_set_proxy (rtspsrc, g_value_get_string (value));
+      break;
+    case PROP_PROXY_ID:
+      if (rtspsrc->prop_proxy_id)
+        g_free (rtspsrc->prop_proxy_id);
+      rtspsrc->prop_proxy_id = g_value_dup_string (value);
+      break;
+    case PROP_PROXY_PW:
+      if (rtspsrc->prop_proxy_pw)
+        g_free (rtspsrc->prop_proxy_pw);
+      rtspsrc->prop_proxy_pw = g_value_dup_string (value);
+      break;
+    case PROP_RTP_BLOCKSIZE:
+      rtspsrc->rtp_blocksize = g_value_get_uint (value);
+      break;
+    case PROP_USER_ID:
+      if (rtspsrc->user_id)
+        g_free (rtspsrc->user_id);
+      rtspsrc->user_id = g_value_dup_string (value);
+      break;
+    case PROP_USER_PW:
+      if (rtspsrc->user_pw)
+        g_free (rtspsrc->user_pw);
+      rtspsrc->user_pw = g_value_dup_string (value);
+      break;
+    case PROP_BUFFER_MODE:
+      rtspsrc->buffer_mode = g_value_get_enum (value);
+      break;
+    case PROP_PORT_RANGE:
+    {
+      const gchar *str;
+
+      str = g_value_get_string (value);
+      if (str) {
+        sscanf (str, "%u-%u",
+            &rtspsrc->client_port_range.min, &rtspsrc->client_port_range.max);
+      } else {
+        rtspsrc->client_port_range.min = 0;
+        rtspsrc->client_port_range.max = 0;
+      }
+      break;
+    }
+    case PROP_UDP_BUFFER_SIZE:
+      rtspsrc->udp_buffer_size = g_value_get_int (value);
+      break;
+    case PROP_SHORT_HEADER:
+      rtspsrc->short_header = g_value_get_boolean (value);
+      break;
+    case PROP_PROBATION:
+      rtspsrc->probation = g_value_get_uint (value);
+      break;
+    case PROP_UDP_RECONNECT:
+      rtspsrc->udp_reconnect = g_value_get_boolean (value);
+      break;
+    case PROP_MULTICAST_IFACE:
+      g_free (rtspsrc->multi_iface);
+
+      if (g_value_get_string (value) == NULL)
+        rtspsrc->multi_iface = g_strdup (DEFAULT_MULTICAST_IFACE);
+      else
+        rtspsrc->multi_iface = g_value_dup_string (value);
+      break;
+    case PROP_NTP_SYNC:
+      rtspsrc->ntp_sync = g_value_get_boolean (value);
+      break;
+    case PROP_USE_PIPELINE_CLOCK:
+      rtspsrc->use_pipeline_clock = g_value_get_boolean (value);
+      break;
+    case PROP_SDES:
+      rtspsrc->sdes = g_value_dup_boxed (value);
+      break;
+    case PROP_TLS_VALIDATION_FLAGS:
+      rtspsrc->tls_validation_flags = g_value_get_flags (value);
+      break;
+    case PROP_TLS_DATABASE:
+      g_clear_object (&rtspsrc->tls_database);
+      rtspsrc->tls_database = g_value_dup_object (value);
+      break;
+    case PROP_DO_RETRANSMISSION:
+      rtspsrc->do_retransmission = g_value_get_boolean (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rtspsrc_get_property (GObject * object, guint prop_id, GValue * value,
+    GParamSpec * pspec)
+{
+  GstRTSPSrc *rtspsrc;
+
+  rtspsrc = GST_RTSPSRC (object);
+
+  switch (prop_id) {
+    case PROP_LOCATION:
+      g_value_set_string (value, rtspsrc->conninfo.location);
+      break;
+    case PROP_PROTOCOLS:
+      g_value_set_flags (value, rtspsrc->protocols);
+      break;
+    case PROP_DEBUG:
+      g_value_set_boolean (value, rtspsrc->debug);
+      break;
+    case PROP_RETRY:
+      g_value_set_uint (value, rtspsrc->retry);
+      break;
+    case PROP_TIMEOUT:
+      g_value_set_uint64 (value, rtspsrc->udp_timeout);
+      break;
+#ifdef GST_EXT_RTSP_MODIFICATION
+    case PROP_START_POSITION:
+      g_value_set_uint64 (value, rtspsrc->start_position);
+      break;
+#endif
+    case PROP_TCP_TIMEOUT:
+    {
+      guint64 timeout;
+
+      timeout = rtspsrc->tcp_timeout.tv_sec * G_USEC_PER_SEC +
+          rtspsrc->tcp_timeout.tv_usec;
+      g_value_set_uint64 (value, timeout);
+      break;
+    }
+    case PROP_LATENCY:
+      g_value_set_uint (value, rtspsrc->latency);
+      break;
+    case PROP_DROP_ON_LATENCY:
+      g_value_set_boolean (value, rtspsrc->drop_on_latency);
+      break;
+    case PROP_CONNECTION_SPEED:
+      g_value_set_uint64 (value, rtspsrc->connection_speed);
+      break;
+    case PROP_NAT_METHOD:
+      g_value_set_enum (value, rtspsrc->nat_method);
+      break;
+    case PROP_DO_RTCP:
+      g_value_set_boolean (value, rtspsrc->do_rtcp);
+      break;
+    case PROP_DO_RTSP_KEEP_ALIVE:
+      g_value_set_boolean (value, rtspsrc->do_rtsp_keep_alive);
+      break;
+    case PROP_PROXY:
+    {
+      gchar *str;
+
+      if (rtspsrc->proxy_host) {
+        str =
+            g_strdup_printf ("%s:%d", rtspsrc->proxy_host, rtspsrc->proxy_port);
+      } else {
+        str = NULL;
+      }
+      g_value_take_string (value, str);
+      break;
+    }
+    case PROP_PROXY_ID:
+      g_value_set_string (value, rtspsrc->prop_proxy_id);
+      break;
+    case PROP_PROXY_PW:
+      g_value_set_string (value, rtspsrc->prop_proxy_pw);
+      break;
+    case PROP_RTP_BLOCKSIZE:
+      g_value_set_uint (value, rtspsrc->rtp_blocksize);
+      break;
+    case PROP_USER_ID:
+      g_value_set_string (value, rtspsrc->user_id);
+      break;
+    case PROP_USER_PW:
+      g_value_set_string (value, rtspsrc->user_pw);
+      break;
+    case PROP_BUFFER_MODE:
+      g_value_set_enum (value, rtspsrc->buffer_mode);
+      break;
+    case PROP_PORT_RANGE:
+    {
+      gchar *str;
+
+      if (rtspsrc->client_port_range.min != 0) {
+        str = g_strdup_printf ("%u-%u", rtspsrc->client_port_range.min,
+            rtspsrc->client_port_range.max);
+      } else {
+        str = NULL;
+      }
+      g_value_take_string (value, str);
+      break;
+    }
+    case PROP_UDP_BUFFER_SIZE:
+      g_value_set_int (value, rtspsrc->udp_buffer_size);
+      break;
+    case PROP_SHORT_HEADER:
+      g_value_set_boolean (value, rtspsrc->short_header);
+      break;
+    case PROP_PROBATION:
+      g_value_set_uint (value, rtspsrc->probation);
+      break;
+    case PROP_UDP_RECONNECT:
+      g_value_set_boolean (value, rtspsrc->udp_reconnect);
+      break;
+    case PROP_MULTICAST_IFACE:
+      g_value_set_string (value, rtspsrc->multi_iface);
+      break;
+    case PROP_NTP_SYNC:
+      g_value_set_boolean (value, rtspsrc->ntp_sync);
+      break;
+    case PROP_USE_PIPELINE_CLOCK:
+      g_value_set_boolean (value, rtspsrc->use_pipeline_clock);
+      break;
+    case PROP_SDES:
+      g_value_set_boxed (value, rtspsrc->sdes);
+      break;
+    case PROP_TLS_VALIDATION_FLAGS:
+      g_value_set_flags (value, rtspsrc->tls_validation_flags);
+      break;
+    case PROP_TLS_DATABASE:
+      g_value_set_object (value, rtspsrc->tls_database);
+      break;
+    case PROP_DO_RETRANSMISSION:
+      g_value_set_boolean (value, rtspsrc->do_retransmission);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static gint
+find_stream_by_id (GstRTSPStream * stream, gint * id)
+{
+  if (stream->id == *id)
+    return 0;
+
+  return -1;
+}
+
+static gint
+find_stream_by_channel (GstRTSPStream * stream, gint * channel)
+{
+  if (stream->channel[0] == *channel || stream->channel[1] == *channel)
+    return 0;
+
+  return -1;
+}
+
+static gint
+find_stream_by_udpsrc (GstRTSPStream * stream, gconstpointer a)
+{
+  GstElement *src = (GstElement *) a;
+
+  if (stream->udpsrc[0] == src)
+    return 0;
+  if (stream->udpsrc[1] == src)
+    return 0;
+
+  return -1;
+}
+
+static gint
+find_stream_by_setup (GstRTSPStream * stream, gconstpointer a)
+{
+  if (stream->conninfo.location) {
+    /* check qualified setup_url */
+    if (!strcmp (stream->conninfo.location, (gchar *) a))
+      return 0;
+  }
+  if (stream->control_url) {
+    /* check original control_url */
+    if (!strcmp (stream->control_url, (gchar *) a))
+      return 0;
+
+    /* check if qualified setup_url ends with string */
+    if (g_str_has_suffix (stream->control_url, (gchar *) a))
+      return 0;
+  }
+
+  return -1;
+}
+
+static GstRTSPStream *
+find_stream (GstRTSPSrc * src, gconstpointer data, gconstpointer func)
+{
+  GList *lstream;
+
+  /* find and get stream */
+  if ((lstream = g_list_find_custom (src->streams, data, (GCompareFunc) func)))
+    return (GstRTSPStream *) lstream->data;
+
+  return NULL;
+}
+
+static const GstSDPBandwidth *
+gst_rtspsrc_get_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
+    const GstSDPMedia * media, const gchar * type)
+{
+  guint i, len;
+
+  /* first look in the media specific section */
+  len = gst_sdp_media_bandwidths_len (media);
+  for (i = 0; i < len; i++) {
+    const GstSDPBandwidth *bw = gst_sdp_media_get_bandwidth (media, i);
+
+    if (strcmp (bw->bwtype, type) == 0)
+      return bw;
+  }
+  /* then look in the message specific section */
+  len = gst_sdp_message_bandwidths_len (sdp);
+  for (i = 0; i < len; i++) {
+    const GstSDPBandwidth *bw = gst_sdp_message_get_bandwidth (sdp, i);
+
+    if (strcmp (bw->bwtype, type) == 0)
+      return bw;
+  }
+  return NULL;
+}
+
+static void
+gst_rtspsrc_collect_bandwidth (GstRTSPSrc * src, const GstSDPMessage * sdp,
+    const GstSDPMedia * media, GstRTSPStream * stream)
+{
+  const GstSDPBandwidth *bw;
+
+  if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_AS)))
+    stream->as_bandwidth = bw->bandwidth;
+  else
+    stream->as_bandwidth = -1;
+
+  if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RR)))
+    stream->rr_bandwidth = bw->bandwidth;
+  else
+    stream->rr_bandwidth = -1;
+
+  if ((bw = gst_rtspsrc_get_bandwidth (src, sdp, media, GST_SDP_BWTYPE_RS)))
+    stream->rs_bandwidth = bw->bandwidth;
+  else
+    stream->rs_bandwidth = -1;
+}
+
+static void
+gst_rtspsrc_do_stream_connection (GstRTSPSrc * src, GstRTSPStream * stream,
+    const GstSDPConnection * conn)
+{
+  if (conn->nettype == NULL || strcmp (conn->nettype, "IN") != 0)
+    return;
+
+  if (conn->addrtype == NULL)
+    return;
+
+  /* check for IPV6 */
+  if (strcmp (conn->addrtype, "IP4") == 0)
+    stream->is_ipv6 = FALSE;
+  else if (strcmp (conn->addrtype, "IP6") == 0)
+    stream->is_ipv6 = TRUE;
+  else
+    return;
+
+  /* save address */
+  g_free (stream->destination);
+  stream->destination = g_strdup (conn->address);
+
+  /* check for multicast */
+  stream->is_multicast =
+      gst_sdp_address_is_multicast (conn->nettype, conn->addrtype,
+      conn->address);
+  stream->ttl = conn->ttl;
+}
+
+/* Go over the connections for a stream.
+ * - If we are dealing with IPV6, we will setup IPV6 sockets for sending and
+ *   receiving.
+ * - If we are dealing with a localhost address, we disable multicast
+ */
+static void
+gst_rtspsrc_collect_connections (GstRTSPSrc * src, const GstSDPMessage * sdp,
+    const GstSDPMedia * media, GstRTSPStream * stream)
+{
+  const GstSDPConnection *conn;
+  guint i, len;
+
+  /* first look in the media specific section */
+  len = gst_sdp_media_connections_len (media);
+  for (i = 0; i < len; i++) {
+    conn = gst_sdp_media_get_connection (media, i);
+
+    gst_rtspsrc_do_stream_connection (src, stream, conn);
+  }
+  /* then look in the message specific section */
+  if ((conn = gst_sdp_message_get_connection (sdp))) {
+    gst_rtspsrc_do_stream_connection (src, stream, conn);
+  }
+}
+
+/*   m=<media> <UDP port> RTP/AVP <payload>
+ */
+static void
+gst_rtspsrc_collect_payloads (GstRTSPSrc * src, const GstSDPMessage * sdp,
+    const GstSDPMedia * media, GstRTSPStream * stream)
+{
+  guint i, len;
+  const gchar *proto;
+
+  /* get proto */
+  proto = gst_sdp_media_get_proto (media);
+  if (proto == NULL)
+    goto no_proto;
+
+  if (g_str_equal (proto, "RTP/AVP"))
+    stream->profile = GST_RTSP_PROFILE_AVP;
+  else if (g_str_equal (proto, "RTP/SAVP"))
+    stream->profile = GST_RTSP_PROFILE_SAVP;
+  else if (g_str_equal (proto, "RTP/AVPF"))
+    stream->profile = GST_RTSP_PROFILE_AVPF;
+  else if (g_str_equal (proto, "RTP/SAVPF"))
+    stream->profile = GST_RTSP_PROFILE_SAVPF;
+  else
+    goto unknown_proto;
+
+  len = gst_sdp_media_formats_len (media);
+  for (i = 0; i < len; i++) {
+    gint pt;
+    GstCaps *caps;
+    GstStructure *s;
+    const gchar *enc;
+    PtMapItem item;
+
+    pt = atoi (gst_sdp_media_get_format (media, i));
+
+    GST_DEBUG_OBJECT (src, " looking at %d pt: %d", i, pt);
+
+    /* convert caps */
+    caps = gst_rtspsrc_media_to_caps (pt, media);
+    if (caps == NULL) {
+      GST_WARNING_OBJECT (src, " skipping pt %d without caps", pt);
+      continue;
+    }
+
+    /* do some tweaks */
+    s = gst_caps_get_structure (caps, 0);
+    if ((enc = gst_structure_get_string (s, "encoding-name"))) {
+      stream->is_real = (strstr (enc, "-REAL") != NULL);
+      if (strcmp (enc, "X-ASF-PF") == 0)
+        stream->container = TRUE;
+    }
+    GST_DEBUG ("mapping sdp session level attributes to caps");
+    gst_rtspsrc_sdp_attributes_to_caps (sdp->attributes, caps);
+    GST_DEBUG ("mapping sdp media level attributes to caps");
+    gst_rtspsrc_sdp_attributes_to_caps (media->attributes, caps);
+
+    /* the first pt will be the default */
+    if (stream->ptmap->len == 0)
+      stream->default_pt = pt;
+
+    item.pt = pt;
+    item.caps = caps;
+    g_array_append_val (stream->ptmap, item);
+  }
+  return;
+
+no_proto:
+  {
+    GST_ERROR_OBJECT (src, "can't find proto in media");
+    return;
+  }
+unknown_proto:
+  {
+    GST_ERROR_OBJECT (src, "unknown proto in media %s", proto);
+    return;
+  }
+}
+
+static const gchar *
+get_aggregate_control (GstRTSPSrc * src)
+{
+  const gchar *base;
+
+  if (src->control)
+    base = src->control;
+  else if (src->content_base)
+    base = src->content_base;
+  else if (src->conninfo.url_str)
+    base = src->conninfo.url_str;
+  else
+    base = "/";
+
+  return base;
+}
+
+static void
+clear_ptmap_item (PtMapItem * item)
+{
+  if (item->caps)
+    gst_caps_unref (item->caps);
+}
+
+static GstRTSPStream *
+gst_rtspsrc_create_stream (GstRTSPSrc * src, GstSDPMessage * sdp, gint idx)
+{
+  GstRTSPStream *stream;
+  const gchar *control_url;
+  const GstSDPMedia *media;
+
+  /* get media, should not return NULL */
+  media = gst_sdp_message_get_media (sdp, idx);
+  if (media == NULL)
+    return NULL;
+
+  stream = g_new0 (GstRTSPStream, 1);
+  stream->parent = src;
+  /* we mark the pad as not linked, we will mark it as OK when we add the pad to
+   * the element. */
+  stream->last_ret = GST_FLOW_NOT_LINKED;
+  stream->added = FALSE;
+  stream->setup = FALSE;
+  stream->skipped = FALSE;
+  stream->id = idx;
+  stream->eos = FALSE;
+  stream->discont = TRUE;
+  stream->seqbase = -1;
+  stream->timebase = -1;
+  stream->send_ssrc = g_random_int ();
+  stream->profile = GST_RTSP_PROFILE_AVP;
+  stream->ptmap = g_array_new (FALSE, FALSE, sizeof (PtMapItem));
+  g_array_set_clear_func (stream->ptmap, (GDestroyNotify) clear_ptmap_item);
+
+  /* collect bandwidth information for this steam. FIXME, configure in the RTP
+   * session manager to scale RTCP. */
+  gst_rtspsrc_collect_bandwidth (src, sdp, media, stream);
+
+  /* collect connection info */
+  gst_rtspsrc_collect_connections (src, sdp, media, stream);
+
+  /* make the payload type map */
+  gst_rtspsrc_collect_payloads (src, sdp, media, stream);
+
+  /* collect port number */
+  stream->port = gst_sdp_media_get_port (media);
+
+  /* get control url to construct the setup url. The setup url is used to
+   * configure the transport of the stream and is used to identity the stream in
+   * the RTP-Info header field returned from PLAY. */
+  control_url = gst_sdp_media_get_attribute_val (media, "control");
+  if (control_url == NULL)
+    control_url = gst_sdp_message_get_attribute_val_n (sdp, "control", 0);
+
+  GST_DEBUG_OBJECT (src, "stream %d, (%p)", stream->id, stream);
+  GST_DEBUG_OBJECT (src, " port: %d", stream->port);
+  GST_DEBUG_OBJECT (src, " container: %d", stream->container);
+  GST_DEBUG_OBJECT (src, " control: %s", GST_STR_NULL (control_url));
+
+  if (control_url != NULL) {
+    stream->control_url = g_strdup (control_url);
+    /* Build a fully qualified url using the content_base if any or by prefixing
+     * the original request.
+     * If the control_url starts with a '/' or a non rtsp: protocol we will most
+     * likely build a URL that the server will fail to understand, this is ok,
+     * we will fail then. */
+    if (g_str_has_prefix (control_url, "rtsp://"))
+      stream->conninfo.location = g_strdup (control_url);
+    else {
+      const gchar *base;
+      gboolean has_slash;
+
+      if (g_strcmp0 (control_url, "*") == 0)
+        control_url = "";
+
+      base = get_aggregate_control (src);
+
+      /* check if the base ends or control starts with / */
+      has_slash = g_str_has_prefix (control_url, "/");
+      has_slash = has_slash || g_str_has_suffix (base, "/");
+
+      /* concatenate the two strings, insert / when not present */
+      stream->conninfo.location =
+          g_strdup_printf ("%s%s%s", base, has_slash ? "" : "/", control_url);
+    }
+  }
+  GST_DEBUG_OBJECT (src, " setup: %s",
+      GST_STR_NULL (stream->conninfo.location));
+
+  /* we keep track of all streams */
+  src->streams = g_list_append (src->streams, stream);
+
+  return stream;
+
+  /* ERRORS */
+}
+
+static void
+gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
+{
+  gint i;
+
+  GST_DEBUG_OBJECT (src, "free stream %p", stream);
+
+  g_array_free (stream->ptmap, TRUE);
+
+  g_free (stream->destination);
+  g_free (stream->control_url);
+  g_free (stream->conninfo.location);
+
+  for (i = 0; i < 2; i++) {
+    if (stream->udpsrc[i]) {
+      gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
+      gst_bin_remove (GST_BIN_CAST (src), stream->udpsrc[i]);
+      gst_object_unref (stream->udpsrc[i]);
+    }
+    if (stream->channelpad[i])
+      gst_object_unref (stream->channelpad[i]);
+
+    if (stream->udpsink[i]) {
+      gst_element_set_state (stream->udpsink[i], GST_STATE_NULL);
+      gst_bin_remove (GST_BIN_CAST (src), stream->udpsink[i]);
+      gst_object_unref (stream->udpsink[i]);
+    }
+  }
+  if (stream->fakesrc) {
+    gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
+    gst_bin_remove (GST_BIN_CAST (src), stream->fakesrc);
+    gst_object_unref (stream->fakesrc);
+  }
+  if (stream->srcpad) {
+    gst_pad_set_active (stream->srcpad, FALSE);
+    if (stream->added)
+      gst_element_remove_pad (GST_ELEMENT_CAST (src), stream->srcpad);
+  }
+  if (stream->srtpenc)
+    gst_object_unref (stream->srtpenc);
+  if (stream->srtpdec)
+    gst_object_unref (stream->srtpdec);
+  if (stream->srtcpparams)
+    gst_caps_unref (stream->srtcpparams);
+  if (stream->rtcppad)
+    gst_object_unref (stream->rtcppad);
+  if (stream->session)
+    g_object_unref (stream->session);
+  if (stream->rtx_pt_map)
+    gst_structure_free (stream->rtx_pt_map);
+  g_free (stream);
+}
+
+static void
+gst_rtspsrc_cleanup (GstRTSPSrc * src)
+{
+  GList *walk;
+
+  GST_DEBUG_OBJECT (src, "cleanup");
+
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+
+    gst_rtspsrc_stream_free (src, stream);
+  }
+  g_list_free (src->streams);
+  src->streams = NULL;
+  if (src->manager) {
+    if (src->manager_sig_id) {
+      g_signal_handler_disconnect (src->manager, src->manager_sig_id);
+      src->manager_sig_id = 0;
+    }
+    gst_element_set_state (src->manager, GST_STATE_NULL);
+    gst_bin_remove (GST_BIN_CAST (src), src->manager);
+    src->manager = NULL;
+  }
+  if (src->props)
+    gst_structure_free (src->props);
+  src->props = NULL;
+
+  g_free (src->content_base);
+  src->content_base = NULL;
+
+  g_free (src->control);
+  src->control = NULL;
+
+  if (src->range)
+    gst_rtsp_range_free (src->range);
+  src->range = NULL;
+
+  /* don't clear the SDP when it was used in the url */
+  if (src->sdp && !src->from_sdp) {
+    gst_sdp_message_free (src->sdp);
+    src->sdp = NULL;
+  }
+
+  src->need_segment = FALSE;
+
+  if (src->provided_clock) {
+    gst_object_unref (src->provided_clock);
+    src->provided_clock = NULL;
+  }
+}
+
+#define PARSE_INT(p, del, res)          \
+G_STMT_START {                          \
+  gchar *t = p;                         \
+  p = strstr (p, del);                  \
+  if (p == NULL)                        \
+    res = -1;                           \
+  else {                                \
+    *p = '\0';                          \
+    p++;                                \
+    res = atoi (t);                     \
+  }                                     \
+} G_STMT_END
+
+#define PARSE_STRING(p, del, res)       \
+G_STMT_START {                          \
+  gchar *t = p;                         \
+  p = strstr (p, del);                  \
+  if (p == NULL) {                      \
+    res = NULL;                         \
+    p = t;                              \
+  }                                     \
+  else {                                \
+    *p = '\0';                          \
+    p++;                                \
+    res = t;                            \
+  }                                     \
+} G_STMT_END
+
+#define SKIP_SPACES(p)                  \
+  while (*p && g_ascii_isspace (*p))    \
+    p++;
+
+/* rtpmap contains:
+ *
+ *  <payload> <encoding_name>/<clock_rate>[/<encoding_params>]
+ */
+static gboolean
+gst_rtspsrc_parse_rtpmap (const gchar * rtpmap, gint * payload, gchar ** name,
+    gint * rate, gchar ** params)
+{
+  gchar *p, *t;
+
+  p = (gchar *) rtpmap;
+
+  PARSE_INT (p, " ", *payload);
+  if (*payload == -1)
+    return FALSE;
+
+  SKIP_SPACES (p);
+  if (*p == '\0')
+    return FALSE;
+
+  PARSE_STRING (p, "/", *name);
+  if (*name == NULL) {
+    GST_DEBUG ("no rate, name %s", p);
+    /* no rate, assume -1 then, this is not supposed to happen but RealMedia
+     * streams seem to omit the rate. */
+    *name = p;
+    *rate = -1;
+    return TRUE;
+  }
+
+  t = p;
+  p = strstr (p, "/");
+  if (p == NULL) {
+    *rate = atoi (t);
+    return TRUE;
+  }
+  *p = '\0';
+  p++;
+  *rate = atoi (t);
+
+  t = p;
+  if (*p == '\0')
+    return TRUE;
+  *params = t;
+
+  return TRUE;
+}
+
+static gboolean
+parse_keymgmt (const gchar * keymgmt, GstCaps * caps)
+{
+  gboolean res = FALSE;
+  gchar *p, *kmpid;
+  gsize size;
+  guchar *data;
+  GstMIKEYMessage *msg;
+  const GstMIKEYPayload *payload;
+  const gchar *srtp_cipher;
+  const gchar *srtp_auth;
+
+  p = (gchar *) keymgmt;
+
+  SKIP_SPACES (p);
+  if (*p == '\0')
+    return FALSE;
+
+  PARSE_STRING (p, " ", kmpid);
+  if (!g_str_equal (kmpid, "mikey"))
+    return FALSE;
+
+  data = g_base64_decode (p, &size);
+  if (data == NULL)
+    return FALSE;
+
+  msg = gst_mikey_message_new_from_data (data, size, NULL, NULL);
+  g_free (data);
+  if (msg == NULL)
+    return FALSE;
+
+  srtp_cipher = "aes-128-icm";
+  srtp_auth = "hmac-sha1-80";
+
+  /* check the Security policy if any */
+  if ((payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_SP, 0))) {
+    GstMIKEYPayloadSP *p = (GstMIKEYPayloadSP *) payload;
+    guint len, i;
+
+    if (p->proto != GST_MIKEY_SEC_PROTO_SRTP)
+      goto done;
+
+    len = gst_mikey_payload_sp_get_n_params (payload);
+    for (i = 0; i < len; i++) {
+      const GstMIKEYPayloadSPParam *param =
+          gst_mikey_payload_sp_get_param (payload, i);
+
+      switch (param->type) {
+        case GST_MIKEY_SP_SRTP_ENC_ALG:
+          switch (param->val[0]) {
+            case 0:
+              srtp_cipher = "null";
+              break;
+            case 2:
+            case 1:
+              srtp_cipher = "aes-128-icm";
+              break;
+            default:
+              break;
+          }
+          break;
+        case GST_MIKEY_SP_SRTP_ENC_KEY_LEN:
+          switch (param->val[0]) {
+            case AES_128_KEY_LEN:
+              srtp_cipher = "aes-128-icm";
+              break;
+            case AES_256_KEY_LEN:
+              srtp_cipher = "aes-256-icm";
+              break;
+            default:
+              break;
+          }
+          break;
+        case GST_MIKEY_SP_SRTP_AUTH_ALG:
+          switch (param->val[0]) {
+            case 0:
+              srtp_auth = "null";
+              break;
+            case 2:
+            case 1:
+              srtp_auth = "hmac-sha1-80";
+              break;
+            default:
+              break;
+          }
+          break;
+        case GST_MIKEY_SP_SRTP_AUTH_KEY_LEN:
+          switch (param->val[0]) {
+            case HMAC_32_KEY_LEN:
+              srtp_auth = "hmac-sha1-32";
+              break;
+            case HMAC_80_KEY_LEN:
+              srtp_auth = "hmac-sha1-80";
+              break;
+            default:
+              break;
+          }
+          break;
+        case GST_MIKEY_SP_SRTP_SRTP_ENC:
+          break;
+        case GST_MIKEY_SP_SRTP_SRTCP_ENC:
+          break;
+        default:
+          break;
+      }
+    }
+  }
+
+  if (!(payload = gst_mikey_message_find_payload (msg, GST_MIKEY_PT_KEMAC, 0)))
+    goto done;
+  else {
+    GstMIKEYPayloadKEMAC *p = (GstMIKEYPayloadKEMAC *) payload;
+    const GstMIKEYPayload *sub;
+    GstMIKEYPayloadKeyData *pkd;
+    GstBuffer *buf;
+
+    if (p->enc_alg != GST_MIKEY_ENC_NULL || p->mac_alg != GST_MIKEY_MAC_NULL)
+      goto done;
+
+    if (!(sub = gst_mikey_payload_kemac_get_sub (payload, 0)))
+      goto done;
+
+    if (sub->type != GST_MIKEY_PT_KEY_DATA)
+      goto done;
+
+    pkd = (GstMIKEYPayloadKeyData *) sub;
+    buf =
+        gst_buffer_new_wrapped (g_memdup (pkd->key_data, pkd->key_len),
+        pkd->key_len);
+    gst_caps_set_simple (caps, "srtp-key", GST_TYPE_BUFFER, buf, NULL);
+  }
+
+  gst_caps_set_simple (caps,
+      "srtp-cipher", G_TYPE_STRING, srtp_cipher,
+      "srtp-auth", G_TYPE_STRING, srtp_auth,
+      "srtcp-cipher", G_TYPE_STRING, srtp_cipher,
+      "srtcp-auth", G_TYPE_STRING, srtp_auth, NULL);
+
+  res = TRUE;
+done:
+  gst_mikey_message_unref (msg);
+
+  return res;
+}
+
+/*
+ * Mapping SDP attributes to caps
+ *
+ * prepend 'a-' to IANA registered sdp attributes names
+ * (ie: not prefixed with 'x-') in order to avoid
+ * collision with gstreamer standard caps properties names
+ */
+static void
+gst_rtspsrc_sdp_attributes_to_caps (GArray * attributes, GstCaps * caps)
+{
+  if (attributes->len > 0) {
+    GstStructure *s;
+    guint i;
+
+    s = gst_caps_get_structure (caps, 0);
+
+    for (i = 0; i < attributes->len; i++) {
+      GstSDPAttribute *attr = &g_array_index (attributes, GstSDPAttribute, i);
+      gchar *tofree, *key;
+
+      key = attr->key;
+
+      /* skip some of the attribute we already handle */
+      if (!strcmp (key, "fmtp"))
+        continue;
+      if (!strcmp (key, "rtpmap"))
+        continue;
+      if (!strcmp (key, "control"))
+        continue;
+      if (!strcmp (key, "range"))
+        continue;
+      if (!strcmp (key, "framesize"))
+        continue;
+      if (g_str_equal (key, "key-mgmt")) {
+        parse_keymgmt (attr->value, caps);
+        continue;
+      }
+
+      /* string must be valid UTF8 */
+      if (!g_utf8_validate (attr->value, -1, NULL))
+        continue;
+
+      if (!g_str_has_prefix (key, "x-"))
+        tofree = key = g_strdup_printf ("a-%s", key);
+      else
+        tofree = NULL;
+
+      GST_DEBUG ("adding caps: %s=%s", key, attr->value);
+      gst_structure_set (s, key, G_TYPE_STRING, attr->value, NULL);
+      g_free (tofree);
+    }
+  }
+}
+
+static const gchar *
+rtsp_get_attribute_for_pt (const GstSDPMedia * media, const gchar * name,
+    gint pt)
+{
+  guint i;
+
+  for (i = 0;; i++) {
+    const gchar *attr;
+    gint val;
+
+    if ((attr = gst_sdp_media_get_attribute_val_n (media, name, i)) == NULL)
+      break;
+
+    if (sscanf (attr, "%d ", &val) != 1)
+      continue;
+
+    if (val == pt)
+      return attr;
+  }
+  return NULL;
+}
+
+/*
+ *  Mapping of caps to and from SDP fields:
+ *
+ *   a=rtpmap:<payload> <encoding_name>/<clock_rate>[/<encoding_params>]
+ *   a=framesize:<payload> <width>-<height>
+ *   a=fmtp:<payload> <param>[=<value>];...
+ */
+static GstCaps *
+gst_rtspsrc_media_to_caps (gint pt, const GstSDPMedia * media)
+{
+  GstCaps *caps;
+  const gchar *rtpmap;
+  const gchar *fmtp;
+  const gchar *framesize;
+  gchar *name = NULL;
+  gint rate = -1;
+  gchar *params = NULL;
+  gchar *tmp;
+  GstStructure *s;
+  gint payload = 0;
+  gboolean ret;
+
+  /* get and parse rtpmap */
+  rtpmap = rtsp_get_attribute_for_pt (media, "rtpmap", pt);
+
+  if (rtpmap) {
+    ret = gst_rtspsrc_parse_rtpmap (rtpmap, &payload, &name, &rate, &params);
+    if (!ret) {
+      g_warning ("error parsing rtpmap, ignoring");
+      rtpmap = NULL;
+    }
+  }
+  /* dynamic payloads need rtpmap or we fail */
+  if (rtpmap == NULL && pt >= 96)
+    goto no_rtpmap;
+
+  /* check if we have a rate, if not, we need to look up the rate from the
+   * default rates based on the payload types. */
+  if (rate == -1) {
+    const GstRTPPayloadInfo *info;
+
+    if (GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
+      /* dynamic types, use media and encoding_name */
+      tmp = g_ascii_strdown (media->media, -1);
+      info = gst_rtp_payload_info_for_name (tmp, name);
+      g_free (tmp);
+    } else {
+      /* static types, use payload type */
+      info = gst_rtp_payload_info_for_pt (pt);
+    }
+
+    if (info) {
+      if ((rate = info->clock_rate) == 0)
+        rate = -1;
+    }
+    /* we fail if we cannot find one */
+    if (rate == -1)
+      goto no_rate;
+  }
+
+  tmp = g_ascii_strdown (media->media, -1);
+  caps = gst_caps_new_simple ("application/x-unknown",
+      "media", G_TYPE_STRING, tmp, "payload", G_TYPE_INT, pt, NULL);
+  g_free (tmp);
+  s = gst_caps_get_structure (caps, 0);
+
+  gst_structure_set (s, "clock-rate", G_TYPE_INT, rate, NULL);
+
+  /* encoding name must be upper case */
+  if (name != NULL) {
+    tmp = g_ascii_strup (name, -1);
+    gst_structure_set (s, "encoding-name", G_TYPE_STRING, tmp, NULL);
+    g_free (tmp);
+  }
+
+  /* params must be lower case */
+  if (params != NULL) {
+    tmp = g_ascii_strdown (params, -1);
+    gst_structure_set (s, "encoding-params", G_TYPE_STRING, tmp, NULL);
+    g_free (tmp);
+  }
+
+  /* parse optional fmtp: field */
+  if ((fmtp = rtsp_get_attribute_for_pt (media, "fmtp", pt))) {
+    gchar *p;
+    gint payload = 0;
+
+    p = (gchar *) fmtp;
+
+    /* p is now of the format <payload> <param>[=<value>];... */
+    PARSE_INT (p, " ", payload);
+    if (payload != -1 && payload == pt) {
+      gchar **pairs;
+      gint i;
+
+      /* <param>[=<value>] are separated with ';' */
+      pairs = g_strsplit (p, ";", 0);
+      for (i = 0; pairs[i]; i++) {
+        gchar *valpos;
+        const gchar *val, *key;
+
+        /* the key may not have a '=', the value can have other '='s */
+        valpos = strstr (pairs[i], "=");
+        if (valpos) {
+          /* we have a '=' and thus a value, remove the '=' with \0 */
+          *valpos = '\0';
+          /* value is everything between '=' and ';'. We split the pairs at ;
+           * boundaries so we can take the remainder of the value. Some servers
+           * put spaces around the value which we strip off here. Alternatively
+           * we could strip those spaces in the depayloaders should these spaces
+           * actually carry any meaning in the future. */
+          val = g_strstrip (valpos + 1);
+        } else {
+          /* simple <param>;.. is translated into <param>=1;... */
+          val = "1";
+        }
+        /* strip the key of spaces, convert key to lowercase but not the value. */
+        key = g_strstrip (pairs[i]);
+        if (strlen (key) > 1) {
+          tmp = g_ascii_strdown (key, -1);
+          gst_structure_set (s, tmp, G_TYPE_STRING, val, NULL);
+          g_free (tmp);
+        }
+      }
+      g_strfreev (pairs);
+    }
+  }
+
+  /* parse framesize: field */
+  if ((framesize = gst_sdp_media_get_attribute_val (media, "framesize"))) {
+    gchar *p;
+
+    /* p is now of the format <payload> <width>-<height> */
+    p = (gchar *) framesize;
+
+    PARSE_INT (p, " ", payload);
+    if (payload != -1 && payload == pt) {
+      gst_structure_set (s, "a-framesize", G_TYPE_STRING, p, NULL);
+    }
+  }
+  return caps;
+
+  /* ERRORS */
+no_rtpmap:
+  {
+    g_warning ("rtpmap type not given for dynamic payload %d", pt);
+    return NULL;
+  }
+no_rate:
+  {
+    g_warning ("rate unknown for payload type %d", pt);
+    return NULL;
+  }
+}
+
+static gboolean
+gst_rtspsrc_alloc_udp_ports (GstRTSPStream * stream,
+    gint * rtpport, gint * rtcpport)
+{
+  GstRTSPSrc *src;
+  GstStateChangeReturn ret;
+  GstElement *udpsrc0, *udpsrc1;
+  gint tmp_rtp, tmp_rtcp;
+  guint count;
+  const gchar *host;
+
+  src = stream->parent;
+
+  udpsrc0 = NULL;
+  udpsrc1 = NULL;
+  count = 0;
+
+  /* Start at next port */
+  tmp_rtp = src->next_port_num;
+
+  if (stream->is_ipv6)
+    host = "udp://[::0]";
+  else
+    host = "udp://0.0.0.0";
+
+  /* try to allocate 2 UDP ports, the RTP port should be an even
+   * number and the RTCP port should be the next (uneven) port */
+again:
+
+  if (tmp_rtp != 0 && src->client_port_range.max > 0 &&
+      tmp_rtp >= src->client_port_range.max)
+    goto no_ports;
+
+  udpsrc0 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
+  if (udpsrc0 == NULL)
+    goto no_udp_protocol;
+  g_object_set (G_OBJECT (udpsrc0), "port", tmp_rtp, "reuse", FALSE, NULL);
+
+  if (src->udp_buffer_size != 0)
+    g_object_set (G_OBJECT (udpsrc0), "buffer-size", src->udp_buffer_size,
+        NULL);
+
+  ret = gst_element_set_state (udpsrc0, GST_STATE_READY);
+  if (ret == GST_STATE_CHANGE_FAILURE) {
+    if (tmp_rtp != 0) {
+      GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTP port %d", tmp_rtp);
+
+      tmp_rtp += 2;
+      if (++count > src->retry)
+        goto no_ports;
+
+      GST_DEBUG_OBJECT (src, "free RTP udpsrc");
+      gst_element_set_state (udpsrc0, GST_STATE_NULL);
+      gst_object_unref (udpsrc0);
+      udpsrc0 = NULL;
+
+      GST_DEBUG_OBJECT (src, "retry %d", count);
+      goto again;
+    }
+    goto no_udp_protocol;
+  }
+
+  g_object_get (G_OBJECT (udpsrc0), "port", &tmp_rtp, NULL);
+  GST_DEBUG_OBJECT (src, "got RTP port %d", tmp_rtp);
+
+  /* check if port is even */
+  if ((tmp_rtp & 0x01) != 0) {
+    /* port not even, close and allocate another */
+    if (++count > src->retry)
+      goto no_ports;
+
+    GST_DEBUG_OBJECT (src, "RTP port not even");
+
+    GST_DEBUG_OBJECT (src, "free RTP udpsrc");
+    gst_element_set_state (udpsrc0, GST_STATE_NULL);
+    gst_object_unref (udpsrc0);
+    udpsrc0 = NULL;
+
+    GST_DEBUG_OBJECT (src, "retry %d", count);
+    tmp_rtp++;
+    goto again;
+  }
+
+  /* allocate port+1 for RTCP now */
+  udpsrc1 = gst_element_make_from_uri (GST_URI_SRC, host, NULL, NULL);
+  if (udpsrc1 == NULL)
+    goto no_udp_rtcp_protocol;
+
+  /* set port */
+  tmp_rtcp = tmp_rtp + 1;
+  if (src->client_port_range.max > 0 && tmp_rtcp > src->client_port_range.max)
+    goto no_ports;
+
+  g_object_set (G_OBJECT (udpsrc1), "port", tmp_rtcp, "reuse", FALSE, NULL);
+
+  GST_DEBUG_OBJECT (src, "starting RTCP on port %d", tmp_rtcp);
+  ret = gst_element_set_state (udpsrc1, GST_STATE_READY);
+  /* tmp_rtcp port is busy already : retry to make rtp/rtcp pair */
+  if (ret == GST_STATE_CHANGE_FAILURE) {
+    GST_DEBUG_OBJECT (src, "Unable to make udpsrc from RTCP port %d", tmp_rtcp);
+
+    if (++count > src->retry)
+      goto no_ports;
+
+    GST_DEBUG_OBJECT (src, "free RTP udpsrc");
+    gst_element_set_state (udpsrc0, GST_STATE_NULL);
+    gst_object_unref (udpsrc0);
+    udpsrc0 = NULL;
+
+    GST_DEBUG_OBJECT (src, "free RTCP udpsrc");
+    gst_element_set_state (udpsrc1, GST_STATE_NULL);
+    gst_object_unref (udpsrc1);
+    udpsrc1 = NULL;
+
+    tmp_rtp += 2;
+    GST_DEBUG_OBJECT (src, "retry %d", count);
+    goto again;
+  }
+
+  /* all fine, do port check */
+  g_object_get (G_OBJECT (udpsrc0), "port", rtpport, NULL);
+  g_object_get (G_OBJECT (udpsrc1), "port", rtcpport, NULL);
+
+  /* this should not happen... */
+  if (*rtpport != tmp_rtp || *rtcpport != tmp_rtcp)
+    goto port_error;
+
+  /* we keep these elements, we configure all in configure_transport when the
+   * server told us to really use the UDP ports. */
+  stream->udpsrc[0] = gst_object_ref_sink (udpsrc0);
+  stream->udpsrc[1] = gst_object_ref_sink (udpsrc1);
+  gst_element_set_locked_state (stream->udpsrc[0], TRUE);
+  gst_element_set_locked_state (stream->udpsrc[1], TRUE);
+
+  /* keep track of next available port number when we have a range
+   * configured */
+  if (src->next_port_num != 0)
+    src->next_port_num = tmp_rtcp + 1;
+
+  return TRUE;
+
+  /* ERRORS */
+no_udp_protocol:
+  {
+    GST_DEBUG_OBJECT (src, "could not get UDP source");
+    goto cleanup;
+  }
+no_ports:
+  {
+    GST_DEBUG_OBJECT (src, "could not allocate UDP port pair after %d retries",
+        count);
+    goto cleanup;
+  }
+no_udp_rtcp_protocol:
+  {
+    GST_DEBUG_OBJECT (src, "could not get UDP source for RTCP");
+    goto cleanup;
+  }
+port_error:
+  {
+    GST_DEBUG_OBJECT (src, "ports don't match rtp: %d<->%d, rtcp: %d<->%d",
+        tmp_rtp, *rtpport, tmp_rtcp, *rtcpport);
+    goto cleanup;
+  }
+cleanup:
+  {
+    if (udpsrc0) {
+      gst_element_set_state (udpsrc0, GST_STATE_NULL);
+      gst_object_unref (udpsrc0);
+    }
+    if (udpsrc1) {
+      gst_element_set_state (udpsrc1, GST_STATE_NULL);
+      gst_object_unref (udpsrc1);
+    }
+    return FALSE;
+  }
+}
+
+static void
+gst_rtspsrc_set_state (GstRTSPSrc * src, GstState state)
+{
+  GList *walk;
+
+  if (src->manager)
+    gst_element_set_state (GST_ELEMENT_CAST (src->manager), state);
+
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+    gint i;
+
+    for (i = 0; i < 2; i++) {
+      if (stream->udpsrc[i])
+        gst_element_set_state (stream->udpsrc[i], state);
+    }
+  }
+}
+
+static void
+gst_rtspsrc_flush (GstRTSPSrc * src, gboolean flush, gboolean playing)
+{
+  GstEvent *event;
+  gint cmd;
+  GstState state;
+
+  if (flush) {
+    event = gst_event_new_flush_start ();
+    GST_DEBUG_OBJECT (src, "start flush");
+    cmd = CMD_WAIT;
+    state = GST_STATE_PAUSED;
+  } else {
+    event = gst_event_new_flush_stop (FALSE);
+    GST_DEBUG_OBJECT (src, "stop flush; playing %d", playing);
+    cmd = CMD_LOOP;
+    if (playing)
+      state = GST_STATE_PLAYING;
+    else
+      state = GST_STATE_PAUSED;
+  }
+  gst_rtspsrc_push_event (src, event);
+  gst_rtspsrc_loop_send_cmd (src, cmd, CMD_LOOP);
+  gst_rtspsrc_set_state (src, state);
+}
+
+static GstRTSPResult
+gst_rtspsrc_connection_send (GstRTSPSrc * src, GstRTSPConnection * conn,
+    GstRTSPMessage * message, GTimeVal * timeout)
+{
+  GstRTSPResult ret;
+
+  if (conn)
+    ret = gst_rtsp_connection_send (conn, message, timeout);
+  else
+    ret = GST_RTSP_ERROR;
+
+  return ret;
+}
+
+static GstRTSPResult
+gst_rtspsrc_connection_receive (GstRTSPSrc * src, GstRTSPConnection * conn,
+    GstRTSPMessage * message, GTimeVal * timeout)
+{
+  GstRTSPResult ret;
+
+  if (conn)
+    ret = gst_rtsp_connection_receive (conn, message, timeout);
+  else
+    ret = GST_RTSP_ERROR;
+
+  return ret;
+}
+
+static void
+gst_rtspsrc_get_position (GstRTSPSrc * src)
+{
+  GstQuery *query;
+  GList *walk;
+
+  query = gst_query_new_position (GST_FORMAT_TIME);
+  /*  should be known somewhere down the stream (e.g. jitterbuffer) */
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+    GstFormat fmt;
+    gint64 pos;
+
+    if (stream->srcpad) {
+      if (gst_pad_query (stream->srcpad, query)) {
+        gst_query_parse_position (query, &fmt, &pos);
+        GST_DEBUG_OBJECT (src, "retaining position %" GST_TIME_FORMAT,
+            GST_TIME_ARGS (pos));
+        src->last_pos = pos;
+        goto out;
+      }
+    }
+  }
+
+  src->last_pos = 0;
+
+out:
+
+  gst_query_unref (query);
+}
+
+static gboolean
+gst_rtspsrc_perform_seek (GstRTSPSrc * src, GstEvent * event)
+{
+  gdouble rate;
+  GstFormat format;
+  GstSeekFlags flags;
+  GstSeekType cur_type = GST_SEEK_TYPE_NONE, stop_type;
+  gint64 cur, stop;
+  gboolean flush, skip;
+  gboolean update;
+  gboolean playing;
+  GstSegment seeksegment = { 0, };
+  GList *walk;
+
+  if (event) {
+    GST_DEBUG_OBJECT (src, "doing seek with event");
+
+    gst_event_parse_seek (event, &rate, &format, &flags,
+        &cur_type, &cur, &stop_type, &stop);
+
+    /* no negative rates yet */
+    if (rate < 0.0)
+      goto negative_rate;
+
+    /* we need TIME format */
+    if (format != src->segment.format)
+      goto no_format;
+  } else {
+    GST_DEBUG_OBJECT (src, "doing seek without event");
+    flags = 0;
+    cur_type = GST_SEEK_TYPE_SET;
+    stop_type = GST_SEEK_TYPE_SET;
+  }
+
+  /* get flush flag */
+  flush = flags & GST_SEEK_FLAG_FLUSH;
+  skip = flags & GST_SEEK_FLAG_SKIP;
+
+  /* now we need to make sure the streaming thread is stopped. We do this by
+   * either sending a FLUSH_START event downstream which will cause the
+   * streaming thread to stop with a WRONG_STATE.
+   * For a non-flushing seek we simply pause the task, which will happen as soon
+   * as it completes one iteration (and thus might block when the sink is
+   * blocking in preroll). */
+  if (flush) {
+    GST_DEBUG_OBJECT (src, "starting flush");
+    gst_rtspsrc_flush (src, TRUE, FALSE);
+  } else {
+    if (src->task) {
+      gst_task_pause (src->task);
+    }
+  }
+
+  /* we should now be able to grab the streaming thread because we stopped it
+   * with the above flush/pause code */
+  GST_RTSP_STREAM_LOCK (src);
+
+  GST_DEBUG_OBJECT (src, "stopped streaming");
+
+  /* stop flushing the rtsp connection so we can send PAUSE/PLAY below */
+  gst_rtspsrc_connection_flush (src, FALSE);
+
+  /* copy segment, we need this because we still need the old
+   * segment when we close the current segment. */
+  memcpy (&seeksegment, &src->segment, sizeof (GstSegment));
+
+  /* configure the seek parameters in the seeksegment. We will then have the
+   * right values in the segment to perform the seek */
+  if (event) {
+    GST_DEBUG_OBJECT (src, "configuring seek");
+    gst_segment_do_seek (&seeksegment, rate, format, flags,
+        cur_type, cur, stop_type, stop, &update);
+  }
+
+  /* figure out the last position we need to play. If it's configured (stop !=
+   * -1), use that, else we play until the total duration of the file */
+  if ((stop = seeksegment.stop) == -1)
+    stop = seeksegment.duration;
+
+  playing = (src->state == GST_RTSP_STATE_PLAYING);
+
+  /* if we were playing, pause first */
+  if (playing) {
+    /* obtain current position in case seek fails */
+    gst_rtspsrc_get_position (src);
+    gst_rtspsrc_pause (src, FALSE);
+  }
+  src->skip = skip;
+
+  src->state = GST_RTSP_STATE_SEEKING;
+
+  /* PLAY will add the range header now. */
+  src->need_range = TRUE;
+
+  /* and continue playing */
+  if (playing)
+    gst_rtspsrc_play (src, &seeksegment, FALSE);
+
+  /* prepare for streaming again */
+  if (flush) {
+    /* if we started flush, we stop now */
+    GST_DEBUG_OBJECT (src, "stopping flush");
+    gst_rtspsrc_flush (src, FALSE, playing);
+  }
+
+  /* now we did the seek and can activate the new segment values */
+  memcpy (&src->segment, &seeksegment, sizeof (GstSegment));
+
+  /* if we're doing a segment seek, post a SEGMENT_START message */
+  if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
+    gst_element_post_message (GST_ELEMENT_CAST (src),
+        gst_message_new_segment_start (GST_OBJECT_CAST (src),
+            src->segment.format, src->segment.position));
+  }
+
+  /* now create the newsegment */
+  GST_DEBUG_OBJECT (src, "Creating newsegment from %" G_GINT64_FORMAT
+      " to %" G_GINT64_FORMAT, src->segment.position, stop);
+
+  /* mark discont */
+  GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+    stream->discont = TRUE;
+  }
+
+  GST_RTSP_STREAM_UNLOCK (src);
+
+  return TRUE;
+
+  /* ERRORS */
+negative_rate:
+  {
+    GST_DEBUG_OBJECT (src, "negative playback rates are not supported yet.");
+    return FALSE;
+  }
+no_format:
+  {
+    GST_DEBUG_OBJECT (src, "unsupported format given, seek aborted.");
+    return FALSE;
+  }
+}
+
+static gboolean
+gst_rtspsrc_handle_src_event (GstPad * pad, GstObject * parent,
+    GstEvent * event)
+{
+  GstRTSPSrc *src;
+  gboolean res = TRUE;
+  gboolean forward;
+
+  src = GST_RTSPSRC_CAST (parent);
+
+  GST_DEBUG_OBJECT (src, "pad %s:%s received event %s",
+      GST_DEBUG_PAD_NAME (pad), GST_EVENT_TYPE_NAME (event));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_SEEK:
+      res = gst_rtspsrc_perform_seek (src, event);
+      forward = FALSE;
+      break;
+    case GST_EVENT_QOS:
+    case GST_EVENT_NAVIGATION:
+    case GST_EVENT_LATENCY:
+    default:
+      forward = TRUE;
+      break;
+  }
+  if (forward) {
+    GstPad *target;
+
+    if ((target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad)))) {
+      res = gst_pad_send_event (target, event);
+      gst_object_unref (target);
+    } else {
+      gst_event_unref (event);
+    }
+  } else {
+    gst_event_unref (event);
+  }
+
+  return res;
+}
+
+/* this is the final event function we receive on the internal source pad when
+ * we deal with TCP connections */
+static gboolean
+gst_rtspsrc_handle_internal_src_event (GstPad * pad, GstObject * parent,
+    GstEvent * event)
+{
+  gboolean res;
+
+  GST_DEBUG_OBJECT (pad, "received event %s", GST_EVENT_TYPE_NAME (event));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_SEEK:
+    case GST_EVENT_QOS:
+    case GST_EVENT_NAVIGATION:
+    case GST_EVENT_LATENCY:
+    default:
+      gst_event_unref (event);
+      res = TRUE;
+      break;
+  }
+  return res;
+}
+
+/* this is the final query function we receive on the internal source pad when
+ * we deal with TCP connections */
+static gboolean
+gst_rtspsrc_handle_internal_src_query (GstPad * pad, GstObject * parent,
+    GstQuery * query)
+{
+  GstRTSPSrc *src;
+  gboolean res = TRUE;
+
+  src = GST_RTSPSRC_CAST (gst_pad_get_element_private (pad));
+
+  GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
+      GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
+
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_POSITION:
+    {
+      /* no idea */
+      break;
+    }
+    case GST_QUERY_DURATION:
+    {
+      GstFormat format;
+
+      gst_query_parse_duration (query, &format, NULL);
+
+      switch (format) {
+        case GST_FORMAT_TIME:
+          gst_query_set_duration (query, format, src->segment.duration);
+          break;
+        default:
+          res = FALSE;
+          break;
+      }
+      break;
+    }
+    case GST_QUERY_LATENCY:
+    {
+      /* we are live with a min latency of 0 and unlimited max latency, this
+       * result will be updated by the session manager if there is any. */
+      gst_query_set_latency (query, TRUE, 0, -1);
+      break;
+    }
+    default:
+      break;
+  }
+
+  return res;
+}
+
+/* this query is executed on the ghost source pad exposed on rtspsrc. */
+static gboolean
+gst_rtspsrc_handle_src_query (GstPad * pad, GstObject * parent,
+    GstQuery * query)
+{
+  GstRTSPSrc *src;
+  gboolean res = FALSE;
+
+  src = GST_RTSPSRC_CAST (parent);
+
+  GST_DEBUG_OBJECT (src, "pad %s:%s received query %s",
+      GST_DEBUG_PAD_NAME (pad), GST_QUERY_TYPE_NAME (query));
+
+  switch (GST_QUERY_TYPE (query)) {
+    case GST_QUERY_DURATION:
+    {
+      GstFormat format;
+
+      gst_query_parse_duration (query, &format, NULL);
+
+      switch (format) {
+        case GST_FORMAT_TIME:
+          gst_query_set_duration (query, format, src->segment.duration);
+          res = TRUE;
+          break;
+        default:
+          break;
+      }
+      break;
+    }
+    case GST_QUERY_SEEKING:
+    {
+      GstFormat format;
+
+      gst_query_parse_seeking (query, &format, NULL, NULL, NULL);
+      if (format == GST_FORMAT_TIME) {
+        gboolean seekable =
+            src->cur_protocols != GST_RTSP_LOWER_TRANS_UDP_MCAST;
+
+        /* seeking without duration is unlikely */
+        seekable = seekable && src->seekable && src->segment.duration &&
+            GST_CLOCK_TIME_IS_VALID (src->segment.duration);
+
+        gst_query_set_seeking (query, GST_FORMAT_TIME, seekable, 0,
+            src->segment.duration);
+        res = TRUE;
+      }
+      break;
+    }
+    case GST_QUERY_URI:
+    {
+      gchar *uri;
+
+      uri = gst_rtspsrc_uri_get_uri (GST_URI_HANDLER (src));
+      if (uri != NULL) {
+        gst_query_set_uri (query, uri);
+        g_free (uri);
+        res = TRUE;
+      }
+      break;
+    }
+    default:
+    {
+      GstPad *target = gst_ghost_pad_get_target (GST_GHOST_PAD_CAST (pad));
+
+      /* forward the query to the proxy target pad */
+      if (target) {
+        res = gst_pad_query (target, query);
+        gst_object_unref (target);
+      }
+      break;
+    }
+  }
+
+  return res;
+}
+
+/* callback for RTCP messages to be sent to the server when operating in TCP
+ * mode. */
+static GstFlowReturn
+gst_rtspsrc_sink_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
+{
+  GstRTSPSrc *src;
+  GstRTSPStream *stream;
+  GstFlowReturn res = GST_FLOW_OK;
+  GstMapInfo map;
+  guint8 *data;
+  guint size;
+  GstRTSPResult ret;
+  GstRTSPMessage message = { 0 };
+  GstRTSPConnection *conn;
+
+  stream = (GstRTSPStream *) gst_pad_get_element_private (pad);
+  src = stream->parent;
+
+  gst_buffer_map (buffer, &map, GST_MAP_READ);
+  size = map.size;
+  data = map.data;
+
+  gst_rtsp_message_init_data (&message, stream->channel[1]);
+
+  /* lend the body data to the message */
+  gst_rtsp_message_take_body (&message, data, size);
+
+  if (stream->conninfo.connection)
+    conn = stream->conninfo.connection;
+  else
+    conn = src->conninfo.connection;
+
+  GST_DEBUG_OBJECT (src, "sending %u bytes RTCP", size);
+  ret = gst_rtspsrc_connection_send (src, conn, &message, NULL);
+  GST_DEBUG_OBJECT (src, "sent RTCP, %d", ret);
+
+  /* and steal it away again because we will free it when unreffing the
+   * buffer */
+  gst_rtsp_message_steal_body (&message, &data, &size);
+  gst_rtsp_message_unset (&message);
+
+  gst_buffer_unmap (buffer, &map);
+  gst_buffer_unref (buffer);
+
+  return res;
+}
+
+static GstPadProbeReturn
+pad_blocked (GstPad * pad, GstPadProbeInfo * info, gpointer user_data)
+{
+  GstRTSPSrc *src = user_data;
+
+  GST_DEBUG_OBJECT (src, "pad %s:%s blocked, activating streams",
+      GST_DEBUG_PAD_NAME (pad));
+
+  /* activate the streams */
+  GST_OBJECT_LOCK (src);
+  if (!src->need_activate)
+    goto was_ok;
+
+  src->need_activate = FALSE;
+  GST_OBJECT_UNLOCK (src);
+
+  gst_rtspsrc_activate_streams (src);
+
+  return GST_PAD_PROBE_OK;
+
+was_ok:
+  {
+    GST_OBJECT_UNLOCK (src);
+    return GST_PAD_PROBE_OK;
+  }
+}
+
+static gboolean
+copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
+{
+  GstPad *gpad = GST_PAD_CAST (user_data);
+
+  GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
+  gst_pad_store_sticky_event (gpad, *event);
+
+  return TRUE;
+}
+
+/* this callback is called when the session manager generated a new src pad with
+ * payloaded RTP packets. We simply ghost the pad here. */
+static void
+new_manager_pad (GstElement * manager, GstPad * pad, GstRTSPSrc * src)
+{
+  gchar *name;
+  GstPadTemplate *template;
+  gint id, ssrc, pt;
+  GList *ostreams;
+  GstRTSPStream *stream;
+  gboolean all_added;
+
+  GST_DEBUG_OBJECT (src, "got new manager pad %" GST_PTR_FORMAT, pad);
+
+  GST_RTSP_STATE_LOCK (src);
+  /* find stream */
+  name = gst_object_get_name (GST_OBJECT_CAST (pad));
+  if (sscanf (name, "recv_rtp_src_%u_%u_%u", &id, &ssrc, &pt) != 3)
+    goto unknown_stream;
+
+  GST_DEBUG_OBJECT (src, "stream: %u, SSRC %08x, PT %d", id, ssrc, pt);
+
+  stream = find_stream (src, &id, (gpointer) find_stream_by_id);
+  if (stream == NULL)
+    goto unknown_stream;
+
+  /* save SSRC */
+  stream->ssrc = ssrc;
+
+  /* we'll add it later see below */
+  stream->added = TRUE;
+
+  /* check if we added all streams */
+  all_added = TRUE;
+  for (ostreams = src->streams; ostreams; ostreams = g_list_next (ostreams)) {
+    GstRTSPStream *ostream = (GstRTSPStream *) ostreams->data;
+
+    GST_DEBUG_OBJECT (src, "stream %p, container %d, added %d, setup %d",
+        ostream, ostream->container, ostream->added, ostream->setup);
+
+    /* if we find a stream for which we did a setup that is not added, we
+     * need to wait some more */
+    if (ostream->setup && !ostream->added) {
+      all_added = FALSE;
+      break;
+    }
+  }
+  GST_RTSP_STATE_UNLOCK (src);
+
+  /* create a new pad we will use to stream to */
+  template = gst_static_pad_template_get (&rtptemplate);
+  stream->srcpad = gst_ghost_pad_new_from_template (name, pad, template);
+  gst_object_unref (template);
+  g_free (name);
+
+  gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
+  gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
+  gst_pad_set_active (stream->srcpad, TRUE);
+  gst_pad_sticky_events_foreach (pad, copy_sticky_events, stream->srcpad);
+  gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
+
+  if (all_added) {
+    GST_DEBUG_OBJECT (src, "We added all streams");
+    /* when we get here, all stream are added and we can fire the no-more-pads
+     * signal. */
+    gst_element_no_more_pads (GST_ELEMENT_CAST (src));
+  }
+
+  return;
+
+  /* ERRORS */
+unknown_stream:
+  {
+    GST_DEBUG_OBJECT (src, "ignoring unknown stream");
+    GST_RTSP_STATE_UNLOCK (src);
+    g_free (name);
+    return;
+  }
+}
+
+static GstCaps *
+stream_get_caps_for_pt (GstRTSPStream * stream, guint pt)
+{
+  guint i, len;
+
+  len = stream->ptmap->len;
+  for (i = 0; i < len; i++) {
+    PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
+    if (item->pt == pt)
+      return item->caps;
+  }
+  return NULL;
+}
+
+static GstCaps *
+request_pt_map (GstElement * manager, guint session, guint pt, GstRTSPSrc * src)
+{
+  GstRTSPStream *stream;
+  GstCaps *caps;
+
+  GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
+
+  GST_RTSP_STATE_LOCK (src);
+  stream = find_stream (src, &session, (gpointer) find_stream_by_id);
+  if (!stream)
+    goto unknown_stream;
+
+  if ((caps = stream_get_caps_for_pt (stream, pt)))
+    gst_caps_ref (caps);
+  GST_RTSP_STATE_UNLOCK (src);
+
+  return caps;
+
+unknown_stream:
+  {
+    GST_DEBUG_OBJECT (src, "unknown stream %d", session);
+    GST_RTSP_STATE_UNLOCK (src);
+    return NULL;
+  }
+}
+
+static void
+gst_rtspsrc_do_stream_eos (GstRTSPSrc * src, GstRTSPStream * stream)
+{
+  GST_DEBUG_OBJECT (src, "setting stream for session %u to EOS", stream->id);
+  if (stream->eos)
+    goto was_eos;
+
+  stream->eos = TRUE;
+  gst_rtspsrc_stream_push_event (src, stream, gst_event_new_eos ());
+  return;
+
+  /* ERRORS */
+was_eos:
+  {
+    GST_DEBUG_OBJECT (src, "stream for session %u was already EOS", stream->id);
+    return;
+  }
+}
+
+static void
+on_bye_ssrc (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+  GstRTSPSrc *src = stream->parent;
+  guint ssrc;
+
+  g_object_get (source, "ssrc", &ssrc, NULL);
+
+  GST_DEBUG_OBJECT (src, "source %08x, stream %08x, session %u received BYE",
+      ssrc, stream->ssrc, stream->id);
+
+  if (ssrc == stream->ssrc)
+    gst_rtspsrc_do_stream_eos (src, stream);
+}
+
+static void
+on_timeout (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+  GstRTSPSrc *src = stream->parent;
+  guint ssrc;
+
+  g_object_get (source, "ssrc", &ssrc, NULL);
+
+  GST_WARNING_OBJECT (src, "source %08x, stream %08x in session %u timed out",
+      ssrc, stream->ssrc, stream->id);
+
+  if (ssrc == stream->ssrc)
+    gst_rtspsrc_do_stream_eos (src, stream);
+}
+
+static void
+on_npt_stop (GstElement * rtpbin, guint session, guint ssrc, GstRTSPSrc * src)
+{
+  GstRTSPStream *stream;
+
+  GST_DEBUG_OBJECT (src, "source in session %u reached NPT stop", session);
+
+  /* get stream for session */
+  stream = find_stream (src, &session, (gpointer) find_stream_by_id);
+  if (stream) {
+    gst_rtspsrc_do_stream_eos (src, stream);
+  }
+}
+
+static void
+on_ssrc_active (GObject * session, GObject * source, GstRTSPStream * stream)
+{
+  GST_DEBUG_OBJECT (stream->parent, "source in session %u is active",
+      stream->id);
+}
+
+static void
+set_manager_buffer_mode (GstRTSPSrc * src)
+{
+  GObjectClass *klass;
+
+  if (src->manager == NULL)
+    return;
+
+  klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
+
+  if (!g_object_class_find_property (klass, "buffer-mode"))
+    return;
+
+  if (src->buffer_mode != BUFFER_MODE_AUTO) {
+    g_object_set (src->manager, "buffer-mode", src->buffer_mode, NULL);
+
+    return;
+  }
+
+  GST_DEBUG_OBJECT (src,
+      "auto buffering mode, have clock %" GST_PTR_FORMAT, src->provided_clock);
+
+  if (src->provided_clock) {
+    GstClock *clock = gst_element_get_clock (GST_ELEMENT_CAST (src));
+
+    if (clock == src->provided_clock) {
+      GST_DEBUG_OBJECT (src, "selected synced");
+      g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SYNCED, NULL);
+
+      if (clock)
+        gst_object_unref (clock);
+
+      return;
+    }
+
+    /* Otherwise fall-through and use another buffer mode */
+    if (clock)
+      gst_object_unref (clock);
+  }
+
+  GST_DEBUG_OBJECT (src, "auto buffering mode");
+  if (src->use_buffering) {
+    GST_DEBUG_OBJECT (src, "selected buffer");
+    g_object_set (src->manager, "buffer-mode", BUFFER_MODE_BUFFER, NULL);
+  } else {
+    GST_DEBUG_OBJECT (src, "selected slave");
+    g_object_set (src->manager, "buffer-mode", BUFFER_MODE_SLAVE, NULL);
+  }
+}
+
+static GstCaps *
+request_key (GstElement * srtpdec, guint ssrc, GstRTSPStream * stream)
+{
+  GST_DEBUG ("request key %u", ssrc);
+  return gst_caps_ref (stream_get_caps_for_pt (stream, stream->default_pt));
+}
+
+static GstElement *
+request_rtp_decoder (GstElement * rtpbin, guint session, GstRTSPStream * stream)
+{
+  GST_DEBUG ("decoder session %u, stream %p, %d", session, stream, stream->id);
+  if (stream->id != session)
+    return NULL;
+
+  if (stream->profile != GST_RTSP_PROFILE_SAVP &&
+      stream->profile != GST_RTSP_PROFILE_SAVPF)
+    return NULL;
+
+  if (stream->srtpdec == NULL) {
+    gchar *name;
+
+    name = g_strdup_printf ("srtpdec_%u", session);
+    stream->srtpdec = gst_element_factory_make ("srtpdec", name);
+    g_free (name);
+
+    g_signal_connect (stream->srtpdec, "request-key",
+        (GCallback) request_key, stream);
+  }
+  return gst_object_ref (stream->srtpdec);
+}
+
+static GstElement *
+request_rtcp_encoder (GstElement * rtpbin, guint session,
+    GstRTSPStream * stream)
+{
+  gchar *name;
+  GstPad *pad;
+
+  GST_DEBUG ("decoder session %u, stream %p, %d", session, stream, stream->id);
+  if (stream->id != session)
+    return NULL;
+
+  if (stream->profile != GST_RTSP_PROFILE_SAVP &&
+      stream->profile != GST_RTSP_PROFILE_SAVPF)
+    return NULL;
+
+  if (stream->srtpenc == NULL) {
+    GstStructure *s;
+
+    name = g_strdup_printf ("srtpenc_%u", session);
+    stream->srtpenc = gst_element_factory_make ("srtpenc", name);
+    g_free (name);
+
+    /* get RTCP crypto parameters from caps */
+    s = gst_caps_get_structure (stream->srtcpparams, 0);
+    if (s) {
+      GstBuffer *buf;
+      const gchar *str;
+      GType ciphertype, authtype;
+      GValue rtcp_cipher = G_VALUE_INIT, rtcp_auth = G_VALUE_INIT;
+
+      ciphertype = g_type_from_name ("GstSrtpCipherType");
+      authtype = g_type_from_name ("GstSrtpAuthType");
+      g_value_init (&rtcp_cipher, ciphertype);
+      g_value_init (&rtcp_auth, authtype);
+
+      str = gst_structure_get_string (s, "srtcp-cipher");
+      gst_value_deserialize (&rtcp_cipher, str);
+      str = gst_structure_get_string (s, "srtcp-auth");
+      gst_value_deserialize (&rtcp_auth, str);
+      gst_structure_get (s, "srtp-key", GST_TYPE_BUFFER, &buf, NULL);
+
+      g_object_set_property (G_OBJECT (stream->srtpenc), "rtcp-cipher",
+          &rtcp_cipher);
+      g_object_set_property (G_OBJECT (stream->srtpenc), "rtcp-auth",
+          &rtcp_auth);
+      g_object_set (stream->srtpenc, "key", buf, NULL);
+
+      g_value_unset (&rtcp_cipher);
+      g_value_unset (&rtcp_auth);
+      gst_buffer_unref (buf);
+    }
+  }
+  name = g_strdup_printf ("rtcp_sink_%d", session);
+  pad = gst_element_get_request_pad (stream->srtpenc, name);
+  g_free (name);
+  gst_object_unref (pad);
+
+  return gst_object_ref (stream->srtpenc);
+}
+
+static GstElement *
+request_aux_receiver (GstElement * rtpbin, guint sessid, GstRTSPSrc * src)
+{
+  GstElement *rtx, *bin;
+  GstPad *pad;
+  gchar *name;
+  GstRTSPStream *stream;
+
+  stream = find_stream (src, &sessid, (gpointer) find_stream_by_id);
+  if (!stream) {
+    GST_WARNING_OBJECT (src, "Stream %u not found", sessid);
+    return NULL;
+  }
+
+  GST_INFO_OBJECT (src, "creating retransmision receiver for session %u "
+      "with map %" GST_PTR_FORMAT, sessid, stream->rtx_pt_map);
+  bin = gst_bin_new (NULL);
+  rtx = gst_element_factory_make ("rtprtxreceive", NULL);
+  g_object_set (rtx, "payload-type-map", stream->rtx_pt_map, NULL);
+  gst_bin_add (GST_BIN (bin), rtx);
+
+  pad = gst_element_get_static_pad (rtx, "src");
+  name = g_strdup_printf ("src_%u", sessid);
+  gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
+  g_free (name);
+  gst_object_unref (pad);
+
+  pad = gst_element_get_static_pad (rtx, "sink");
+  name = g_strdup_printf ("sink_%u", sessid);
+  gst_element_add_pad (bin, gst_ghost_pad_new (name, pad));
+  g_free (name);
+  gst_object_unref (pad);
+
+  return bin;
+}
+
+static void
+add_retransmission (GstRTSPSrc * src, GstRTSPTransport * transport)
+{
+  GList *walk;
+  guint signal_id;
+  gboolean do_retransmission = FALSE;
+
+  if (transport->trans != GST_RTSP_TRANS_RTP)
+    return;
+  if (transport->profile != GST_RTSP_PROFILE_AVPF &&
+      transport->profile != GST_RTSP_PROFILE_SAVPF)
+    return;
+
+  signal_id = g_signal_lookup ("request-aux-receiver",
+      G_OBJECT_TYPE (src->manager));
+  /* there's already something connected */
+  if (g_signal_handler_find (src->manager, G_SIGNAL_MATCH_ID, signal_id, 0,
+          NULL, NULL, NULL) != 0) {
+    GST_DEBUG_OBJECT (src, "Not adding RTX AUX element as "
+        "\"request-aux-receiver\" signal is "
+        "already used by the application");
+    return;
+  }
+
+  /* build the retransmission payload type map */
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+    gboolean do_retransmission_stream = FALSE;
+    int i;
+
+    if (stream->rtx_pt_map)
+      gst_structure_free (stream->rtx_pt_map);
+    stream->rtx_pt_map = gst_structure_new_empty ("application/x-rtp-pt-map");
+
+    for (i = 0; i < stream->ptmap->len; i++) {
+      PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
+      GstStructure *s = gst_caps_get_structure (item->caps, 0);
+      const gchar *encoding;
+
+      /* we only care about RTX streams */
+      if ((encoding = gst_structure_get_string (s, "encoding-name"))
+          && g_strcmp0 (encoding, "RTX") == 0) {
+        const gchar *stream_pt_s;
+        gint rtx_pt;
+
+        if (gst_structure_get_int (s, "payload", &rtx_pt)
+            && (stream_pt_s = gst_structure_get_string (s, "apt"))) {
+
+          if (rtx_pt != 0) {
+            gst_structure_set (stream->rtx_pt_map, stream_pt_s, G_TYPE_UINT,
+                rtx_pt, NULL);
+            do_retransmission_stream = TRUE;
+          }
+        }
+      }
+    }
+
+    if (do_retransmission_stream) {
+      GST_DEBUG_OBJECT (src, "built retransmission payload map for stream "
+          "id %i: %" GST_PTR_FORMAT, stream->id, stream->rtx_pt_map);
+      do_retransmission = TRUE;
+    } else {
+      GST_DEBUG_OBJECT (src, "no retransmission payload map for stream "
+          "id %i", stream->id);
+      gst_structure_free (stream->rtx_pt_map);
+      stream->rtx_pt_map = NULL;
+    }
+  }
+
+  if (do_retransmission) {
+    GST_DEBUG_OBJECT (src, "Enabling retransmissions");
+
+    g_object_set (src->manager, "do-retransmission", TRUE, NULL);
+
+    /* enable RFC4588 retransmission handling by setting rtprtxreceive
+     * as the "aux" element of rtpbin */
+    g_signal_connect (src->manager, "request-aux-receiver",
+        (GCallback) request_aux_receiver, src);
+  } else {
+    GST_DEBUG_OBJECT (src,
+        "Not enabling retransmissions as no stream had a retransmission payload map");
+  }
+}
+
+/* try to get and configure a manager */
+static gboolean
+gst_rtspsrc_stream_configure_manager (GstRTSPSrc * src, GstRTSPStream * stream,
+    GstRTSPTransport * transport)
+{
+  const gchar *manager;
+  gchar *name;
+  GstStateChangeReturn ret;
+
+  /* find a manager */
+  if (gst_rtsp_transport_get_manager (transport->trans, &manager, 0) < 0)
+    goto no_manager;
+
+  if (manager) {
+    GST_DEBUG_OBJECT (src, "using manager %s", manager);
+
+    /* configure the manager */
+    if (src->manager == NULL) {
+      GObjectClass *klass;
+
+      if (!(src->manager = gst_element_factory_make (manager, "manager"))) {
+        /* fallback */
+        if (gst_rtsp_transport_get_manager (transport->trans, &manager, 1) < 0)
+          goto no_manager;
+
+        if (!manager)
+          goto use_no_manager;
+
+        if (!(src->manager = gst_element_factory_make (manager, "manager")))
+          goto manager_failed;
+      }
+
+      /* we manage this element */
+      gst_element_set_locked_state (src->manager, TRUE);
+      gst_bin_add (GST_BIN_CAST (src), src->manager);
+
+      ret = gst_element_set_state (src->manager, GST_STATE_PAUSED);
+      if (ret == GST_STATE_CHANGE_FAILURE)
+        goto start_manager_failure;
+
+      g_object_set (src->manager, "latency", src->latency, NULL);
+
+      klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
+
+      if (g_object_class_find_property (klass, "ntp-sync")) {
+        g_object_set (src->manager, "ntp-sync", src->ntp_sync, NULL);
+      }
+
+      if (g_object_class_find_property (klass, "use-pipeline-clock")) {
+        g_object_set (src->manager, "use-pipeline-clock",
+            src->use_pipeline_clock, NULL);
+      }
+
+      if (src->sdes && g_object_class_find_property (klass, "sdes")) {
+        g_object_set (src->manager, "sdes", src->sdes, NULL);
+      }
+
+      if (g_object_class_find_property (klass, "drop-on-latency")) {
+        g_object_set (src->manager, "drop-on-latency", src->drop_on_latency,
+            NULL);
+      }
+
+      /* buffer mode pauses are handled by adding offsets to buffer times,
+       * but some depayloaders may have a hard time syncing output times
+       * with such input times, e.g. container ones, most notably ASF */
+      /* TODO alternatives are having an event that indicates these shifts,
+       * or having rtsp extensions provide suggestion on buffer mode */
+      /* valid duration implies not likely live pipeline,
+       * so slaving in jitterbuffer does not make much sense
+       * (and might mess things up due to bursts) */
+      if (GST_CLOCK_TIME_IS_VALID (src->segment.duration) &&
+          src->segment.duration && stream->container) {
+        src->use_buffering = TRUE;
+      } else {
+        src->use_buffering = FALSE;
+      }
+
+      set_manager_buffer_mode (src);
+
+      /* connect to signals */
+      GST_DEBUG_OBJECT (src, "connect to signals on session manager, stream %p",
+          stream);
+      src->manager_sig_id =
+          g_signal_connect (src->manager, "pad-added",
+          (GCallback) new_manager_pad, src);
+      src->manager_ptmap_id =
+          g_signal_connect (src->manager, "request-pt-map",
+          (GCallback) request_pt_map, src);
+
+      g_signal_connect (src->manager, "on-npt-stop", (GCallback) on_npt_stop,
+          src);
+
+      g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_NEW_MANAGER], 0,
+          src->manager);
+
+      if (src->do_retransmission)
+        add_retransmission (src, transport);
+    }
+    g_signal_connect (src->manager, "request-rtp-decoder",
+        (GCallback) request_rtp_decoder, stream);
+    g_signal_connect (src->manager, "request-rtcp-decoder",
+        (GCallback) request_rtp_decoder, stream);
+    g_signal_connect (src->manager, "request-rtcp-encoder",
+        (GCallback) request_rtcp_encoder, stream);
+
+    /* we stream directly to the manager, get some pads. Each RTSP stream goes
+     * into a separate RTP session. */
+    name = g_strdup_printf ("recv_rtp_sink_%u", stream->id);
+    stream->channelpad[0] = gst_element_get_request_pad (src->manager, name);
+    g_free (name);
+    name = g_strdup_printf ("recv_rtcp_sink_%u", stream->id);
+    stream->channelpad[1] = gst_element_get_request_pad (src->manager, name);
+    g_free (name);
+
+    /* now configure the bandwidth in the manager */
+    if (g_signal_lookup ("get-internal-session",
+            G_OBJECT_TYPE (src->manager)) != 0) {
+      GObject *rtpsession;
+
+      g_signal_emit_by_name (src->manager, "get-internal-session", stream->id,
+          &rtpsession);
+      if (rtpsession) {
+        GST_INFO_OBJECT (src, "configure bandwidth in session %p", rtpsession);
+
+        stream->session = rtpsession;
+
+        if (stream->as_bandwidth != -1) {
+          GST_INFO_OBJECT (src, "setting AS: %f",
+              (gdouble) (stream->as_bandwidth * 1000));
+          g_object_set (rtpsession, "bandwidth",
+              (gdouble) (stream->as_bandwidth * 1000), NULL);
+        }
+        if (stream->rr_bandwidth != -1) {
+          GST_INFO_OBJECT (src, "setting RR: %u", stream->rr_bandwidth);
+          g_object_set (rtpsession, "rtcp-rr-bandwidth", stream->rr_bandwidth,
+              NULL);
+        }
+        if (stream->rs_bandwidth != -1) {
+          GST_INFO_OBJECT (src, "setting RS: %u", stream->rs_bandwidth);
+          g_object_set (rtpsession, "rtcp-rs-bandwidth", stream->rs_bandwidth,
+              NULL);
+        }
+
+        g_object_set (rtpsession, "probation", src->probation, NULL);
+
+        g_object_set (rtpsession, "internal-ssrc", stream->send_ssrc, NULL);
+
+        g_signal_connect (rtpsession, "on-bye-ssrc", (GCallback) on_bye_ssrc,
+            stream);
+        g_signal_connect (rtpsession, "on-bye-timeout", (GCallback) on_timeout,
+            stream);
+        g_signal_connect (rtpsession, "on-timeout", (GCallback) on_timeout,
+            stream);
+        g_signal_connect (rtpsession, "on-ssrc-active",
+            (GCallback) on_ssrc_active, stream);
+      }
+    }
+  }
+
+use_no_manager:
+  return TRUE;
+
+  /* ERRORS */
+no_manager:
+  {
+    GST_DEBUG_OBJECT (src, "cannot get a session manager");
+    return FALSE;
+  }
+manager_failed:
+  {
+    GST_DEBUG_OBJECT (src, "no session manager element %s found", manager);
+    return FALSE;
+  }
+start_manager_failure:
+  {
+    GST_DEBUG_OBJECT (src, "could not start session manager");
+    return FALSE;
+  }
+}
+
+/* free the UDP sources allocated when negotiating a transport.
+ * This function is called when the server negotiated to a transport where the
+ * UDP sources are not needed anymore, such as TCP or multicast. */
+static void
+gst_rtspsrc_stream_free_udp (GstRTSPStream * stream)
+{
+  gint i;
+
+  for (i = 0; i < 2; i++) {
+    if (stream->udpsrc[i]) {
+      GST_DEBUG ("free UDP source %d for stream %p", i, stream);
+      gst_element_set_state (stream->udpsrc[i], GST_STATE_NULL);
+      gst_object_unref (stream->udpsrc[i]);
+      stream->udpsrc[i] = NULL;
+    }
+  }
+}
+
+/* for TCP, create pads to send and receive data to and from the manager and to
+ * intercept various events and queries
+ */
+static gboolean
+gst_rtspsrc_stream_configure_tcp (GstRTSPSrc * src, GstRTSPStream * stream,
+    GstRTSPTransport * transport, GstPad ** outpad)
+{
+  gchar *name;
+  GstPadTemplate *template;
+  GstPad *pad0, *pad1;
+
+  /* configure for interleaved delivery, nothing needs to be done
+   * here, the loop function will call the chain functions of the
+   * session manager. */
+  stream->channel[0] = transport->interleaved.min;
+  stream->channel[1] = transport->interleaved.max;
+  GST_DEBUG_OBJECT (src, "stream %p on channels %d-%d", stream,
+      stream->channel[0], stream->channel[1]);
+
+  /* we can remove the allocated UDP ports now */
+  gst_rtspsrc_stream_free_udp (stream);
+
+  /* no session manager, send data to srcpad directly */
+  if (!stream->channelpad[0]) {
+    GST_DEBUG_OBJECT (src, "no manager, creating pad");
+
+    /* create a new pad we will use to stream to */
+    name = g_strdup_printf ("stream_%u", stream->id);
+    template = gst_static_pad_template_get (&rtptemplate);
+    stream->channelpad[0] = gst_pad_new_from_template (template, name);
+    gst_object_unref (template);
+    g_free (name);
+
+    /* set caps and activate */
+    gst_pad_use_fixed_caps (stream->channelpad[0]);
+    gst_pad_set_active (stream->channelpad[0], TRUE);
+
+    *outpad = gst_object_ref (stream->channelpad[0]);
+  } else {
+    GST_DEBUG_OBJECT (src, "using manager source pad");
+
+    template = gst_static_pad_template_get (&anysrctemplate);
+
+    /* allocate pads for sending the channel data into the manager */
+    pad0 = gst_pad_new_from_template (template, "internalsrc_0");
+    gst_pad_link_full (pad0, stream->channelpad[0], GST_PAD_LINK_CHECK_NOTHING);
+    gst_object_unref (stream->channelpad[0]);
+    stream->channelpad[0] = pad0;
+    gst_pad_set_event_function (pad0, gst_rtspsrc_handle_internal_src_event);
+    gst_pad_set_query_function (pad0, gst_rtspsrc_handle_internal_src_query);
+    gst_pad_set_element_private (pad0, src);
+    gst_pad_set_active (pad0, TRUE);
+
+    if (stream->channelpad[1]) {
+      /* if we have a sinkpad for the other channel, create a pad and link to the
+       * manager. */
+      pad1 = gst_pad_new_from_template (template, "internalsrc_1");
+      gst_pad_set_event_function (pad1, gst_rtspsrc_handle_internal_src_event);
+      gst_pad_link_full (pad1, stream->channelpad[1],
+          GST_PAD_LINK_CHECK_NOTHING);
+      gst_object_unref (stream->channelpad[1]);
+      stream->channelpad[1] = pad1;
+      gst_pad_set_active (pad1, TRUE);
+    }
+    gst_object_unref (template);
+  }
+  /* setup RTCP transport back to the server if we have to. */
+  if (src->manager && src->do_rtcp) {
+    GstPad *pad;
+
+    template = gst_static_pad_template_get (&anysinktemplate);
+
+    stream->rtcppad = gst_pad_new_from_template (template, "internalsink_0");
+    gst_pad_set_chain_function (stream->rtcppad, gst_rtspsrc_sink_chain);
+    gst_pad_set_element_private (stream->rtcppad, stream);
+    gst_pad_set_active (stream->rtcppad, TRUE);
+
+    /* get session RTCP pad */
+    name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
+    pad = gst_element_get_request_pad (src->manager, name);
+    g_free (name);
+
+    /* and link */
+    if (pad) {
+      gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
+      gst_object_unref (pad);
+    }
+
+    gst_object_unref (template);
+  }
+  return TRUE;
+}
+
+static void
+gst_rtspsrc_get_transport_info (GstRTSPSrc * src, GstRTSPStream * stream,
+    GstRTSPTransport * transport, const gchar ** destination, gint * min,
+    gint * max, guint * ttl)
+{
+  if (transport->lower_transport == GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+    if (destination) {
+      if (!(*destination = transport->destination))
+        *destination = stream->destination;
+    }
+    if (min && max) {
+      /* transport first */
+      *min = transport->port.min;
+      *max = transport->port.max;
+      if (*min == -1 && *max == -1) {
+        /* then try from SDP */
+        if (stream->port != 0) {
+          *min = stream->port;
+          *max = stream->port + 1;
+        }
+      }
+    }
+
+    if (ttl) {
+      if (!(*ttl = transport->ttl))
+        *ttl = stream->ttl;
+    }
+  } else {
+    if (destination) {
+      /* first take the source, then the endpoint to figure out where to send
+       * the RTCP. */
+      if (!(*destination = transport->source)) {
+        if (src->conninfo.connection)
+          *destination = gst_rtsp_connection_get_ip (src->conninfo.connection);
+        else if (stream->conninfo.connection)
+          *destination =
+              gst_rtsp_connection_get_ip (stream->conninfo.connection);
+      }
+    }
+    if (min && max) {
+      /* for unicast we only expect the ports here */
+      *min = transport->server_port.min;
+      *max = transport->server_port.max;
+    }
+  }
+}
+
+/* For multicast create UDP sources and join the multicast group. */
+static gboolean
+gst_rtspsrc_stream_configure_mcast (GstRTSPSrc * src, GstRTSPStream * stream,
+    GstRTSPTransport * transport, GstPad ** outpad)
+{
+  gchar *uri;
+  const gchar *destination;
+  gint min, max;
+
+  GST_DEBUG_OBJECT (src, "creating UDP sources for multicast");
+
+  /* we can remove the allocated UDP ports now */
+  gst_rtspsrc_stream_free_udp (stream);
+
+  gst_rtspsrc_get_transport_info (src, stream, transport, &destination, &min,
+      &max, NULL);
+
+  /* we need a destination now */
+  if (destination == NULL)
+    goto no_destination;
+
+  /* we really need ports now or we won't be able to receive anything at all */
+  if (min == -1 && max == -1)
+    goto no_ports;
+
+  GST_DEBUG_OBJECT (src, "have destination '%s' and ports (%d)-(%d)",
+      destination, min, max);
+
+  /* creating UDP source for RTP */
+  if (min != -1) {
+    uri = g_strdup_printf ("udp://%s:%d", destination, min);
+    stream->udpsrc[0] =
+        gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
+    g_free (uri);
+    if (stream->udpsrc[0] == NULL)
+      goto no_element;
+
+    /* take ownership */
+    gst_object_ref_sink (stream->udpsrc[0]);
+
+    if (src->udp_buffer_size != 0)
+      g_object_set (G_OBJECT (stream->udpsrc[0]), "buffer-size",
+          src->udp_buffer_size, NULL);
+
+    if (src->multi_iface != NULL)
+      g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
+          src->multi_iface, NULL);
+
+    /* change state */
+    gst_element_set_locked_state (stream->udpsrc[0], TRUE);
+    gst_element_set_state (stream->udpsrc[0], GST_STATE_READY);
+  }
+
+  /* creating another UDP source for RTCP */
+  if (max != -1) {
+    GstCaps *caps;
+
+    uri = g_strdup_printf ("udp://%s:%d", destination, max);
+    stream->udpsrc[1] =
+        gst_element_make_from_uri (GST_URI_SRC, uri, NULL, NULL);
+    g_free (uri);
+    if (stream->udpsrc[1] == NULL)
+      goto no_element;
+
+    if (stream->profile == GST_RTSP_PROFILE_SAVP ||
+        stream->profile == GST_RTSP_PROFILE_SAVPF)
+      caps = gst_caps_new_empty_simple ("application/x-srtcp");
+    else
+      caps = gst_caps_new_empty_simple ("application/x-rtcp");
+    g_object_set (stream->udpsrc[1], "caps", caps, NULL);
+    gst_caps_unref (caps);
+
+    /* take ownership */
+    gst_object_ref_sink (stream->udpsrc[1]);
+
+    if (src->multi_iface != NULL)
+      g_object_set (G_OBJECT (stream->udpsrc[0]), "multicast-iface",
+          src->multi_iface, NULL);
+
+    gst_element_set_state (stream->udpsrc[1], GST_STATE_READY);
+  }
+  return TRUE;
+
+  /* ERRORS */
+no_element:
+  {
+    GST_DEBUG_OBJECT (src, "no UDP source element found");
+    return FALSE;
+  }
+no_destination:
+  {
+    GST_DEBUG_OBJECT (src, "no destination found");
+    return FALSE;
+  }
+no_ports:
+  {
+    GST_DEBUG_OBJECT (src, "no ports found");
+    return FALSE;
+  }
+}
+
+/* configure the remainder of the UDP ports */
+static gboolean
+gst_rtspsrc_stream_configure_udp (GstRTSPSrc * src, GstRTSPStream * stream,
+    GstRTSPTransport * transport, GstPad ** outpad)
+{
+  /* we manage the UDP elements now. For unicast, the UDP sources where
+   * allocated in the stream when we suggested a transport. */
+  if (stream->udpsrc[0]) {
+    GstCaps *caps;
+
+    gst_element_set_locked_state (stream->udpsrc[0], TRUE);
+    gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[0]);
+
+    GST_DEBUG_OBJECT (src, "setting up UDP source");
+
+    /* configure a timeout on the UDP port. When the timeout message is
+     * posted, we assume UDP transport is not possible. We reconnect using TCP
+     * if we can. */
+    g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout",
+        src->udp_timeout * 1000, NULL);
+
+    if ((caps = stream_get_caps_for_pt (stream, stream->default_pt)))
+      g_object_set (stream->udpsrc[0], "caps", caps, NULL);
+
+    /* get output pad of the UDP source. */
+    *outpad = gst_element_get_static_pad (stream->udpsrc[0], "src");
+
+    /* save it so we can unblock */
+    stream->blockedpad = *outpad;
+
+    /* configure pad block on the pad. As soon as there is dataflow on the
+     * UDP source, we know that UDP is not blocked by a firewall and we can
+     * configure all the streams to let the application autoplug decoders. */
+    stream->blockid =
+        gst_pad_add_probe (stream->blockedpad,
+        GST_PAD_PROBE_TYPE_BLOCK | GST_PAD_PROBE_TYPE_BUFFER |
+        GST_PAD_PROBE_TYPE_BUFFER_LIST, pad_blocked, src, NULL);
+
+    if (stream->channelpad[0]) {
+      GST_DEBUG_OBJECT (src, "connecting UDP source 0 to manager");
+      /* configure for UDP delivery, we need to connect the UDP pads to
+       * the session plugin. */
+      gst_pad_link_full (*outpad, stream->channelpad[0],
+          GST_PAD_LINK_CHECK_NOTHING);
+      gst_object_unref (*outpad);
+      *outpad = NULL;
+      /* we connected to pad-added signal to get pads from the manager */
+    } else {
+      GST_DEBUG_OBJECT (src, "using UDP src pad as output");
+    }
+  }
+
+  /* RTCP port */
+  if (stream->udpsrc[1]) {
+    GstCaps *caps;
+
+    gst_element_set_locked_state (stream->udpsrc[1], TRUE);
+    gst_bin_add (GST_BIN_CAST (src), stream->udpsrc[1]);
+
+    if (stream->profile == GST_RTSP_PROFILE_SAVP ||
+        stream->profile == GST_RTSP_PROFILE_SAVPF)
+      caps = gst_caps_new_empty_simple ("application/x-srtcp");
+    else
+      caps = gst_caps_new_empty_simple ("application/x-rtcp");
+    g_object_set (stream->udpsrc[1], "caps", caps, NULL);
+    gst_caps_unref (caps);
+
+    if (stream->channelpad[1]) {
+      GstPad *pad;
+
+      GST_DEBUG_OBJECT (src, "connecting UDP source 1 to manager");
+
+      pad = gst_element_get_static_pad (stream->udpsrc[1], "src");
+      gst_pad_link_full (pad, stream->channelpad[1],
+          GST_PAD_LINK_CHECK_NOTHING);
+      gst_object_unref (pad);
+    } else {
+      /* leave unlinked */
+    }
+  }
+  return TRUE;
+}
+
+/* configure the UDP sink back to the server for status reports */
+static gboolean
+gst_rtspsrc_stream_configure_udp_sinks (GstRTSPSrc * src,
+    GstRTSPStream * stream, GstRTSPTransport * transport)
+{
+  GstPad *pad;
+  gint rtp_port, rtcp_port;
+  gboolean do_rtp, do_rtcp;
+  const gchar *destination;
+  gchar *uri, *name;
+  guint ttl = 0;
+  GSocket *socket;
+
+  /* get transport info */
+  gst_rtspsrc_get_transport_info (src, stream, transport, &destination,
+      &rtp_port, &rtcp_port, &ttl);
+
+  /* see what we need to do */
+  do_rtp = (rtp_port != -1);
+  /* it's possible that the server does not want us to send RTCP in which case
+   * the port is -1 */
+  do_rtcp = (rtcp_port != -1 && src->manager != NULL && src->do_rtcp);
+
+  /* we need a destination when we have RTP or RTCP ports */
+  if (destination == NULL && (do_rtp || do_rtcp))
+    goto no_destination;
+
+  /* try to construct the fakesrc to the RTP port of the server to open up any
+   * NAT firewalls */
+  if (do_rtp) {
+    GST_DEBUG_OBJECT (src, "configure RTP UDP sink for %s:%d", destination,
+        rtp_port);
+
+    uri = g_strdup_printf ("udp://%s:%d", destination, rtp_port);
+    stream->udpsink[0] =
+        gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
+    g_free (uri);
+    if (stream->udpsink[0] == NULL)
+      goto no_sink_element;
+
+    /* don't join multicast group, we will have the source socket do that */
+    /* no sync or async state changes needed */
+    g_object_set (G_OBJECT (stream->udpsink[0]), "auto-multicast", FALSE,
+        "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
+    if (ttl > 0)
+      g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
+
+    if (stream->udpsrc[0]) {
+      /* configure socket, we give it the same UDP socket as the udpsrc for RTP
+       * so that NAT firewalls will open a hole for us */
+      g_object_get (G_OBJECT (stream->udpsrc[0]), "used-socket", &socket, NULL);
+      GST_DEBUG_OBJECT (src, "RTP UDP src has sock %p", socket);
+      /* configure socket and make sure udpsink does not close it when shutting
+       * down, it belongs to udpsrc after all. */
+      g_object_set (G_OBJECT (stream->udpsink[0]), "socket", socket,
+          "close-socket", FALSE, NULL);
+      g_object_unref (socket);
+    }
+
+    /* the source for the dummy packets to open up NAT */
+    stream->fakesrc = gst_element_factory_make ("fakesrc", NULL);
+    if (stream->fakesrc == NULL)
+      goto no_fakesrc_element;
+
+    /* random data in 5 buffers, a size of 200 bytes should be fine */
+    g_object_set (G_OBJECT (stream->fakesrc), "filltype", 3, "num-buffers", 5,
+        "sizetype", 2, "sizemax", 200, "silent", TRUE, NULL);
+
+    /* we don't want to consider this a sink */
+    GST_OBJECT_FLAG_UNSET (stream->udpsink[0], GST_ELEMENT_FLAG_SINK);
+
+    /* keep everything locked */
+    gst_element_set_locked_state (stream->udpsink[0], TRUE);
+    gst_element_set_locked_state (stream->fakesrc, TRUE);
+
+    gst_object_ref (stream->udpsink[0]);
+    gst_bin_add (GST_BIN_CAST (src), stream->udpsink[0]);
+    gst_object_ref (stream->fakesrc);
+    gst_bin_add (GST_BIN_CAST (src), stream->fakesrc);
+
+    gst_element_link_pads_full (stream->fakesrc, "src", stream->udpsink[0],
+        "sink", GST_PAD_LINK_CHECK_NOTHING);
+  }
+  if (do_rtcp) {
+    GST_DEBUG_OBJECT (src, "configure RTCP UDP sink for %s:%d", destination,
+        rtcp_port);
+
+    uri = g_strdup_printf ("udp://%s:%d", destination, rtcp_port);
+    stream->udpsink[1] =
+        gst_element_make_from_uri (GST_URI_SINK, uri, NULL, NULL);
+    g_free (uri);
+    if (stream->udpsink[1] == NULL)
+      goto no_sink_element;
+
+    /* don't join multicast group, we will have the source socket do that */
+    /* no sync or async state changes needed */
+    g_object_set (G_OBJECT (stream->udpsink[1]), "auto-multicast", FALSE,
+        "loop", FALSE, "sync", FALSE, "async", FALSE, NULL);
+    if (ttl > 0)
+      g_object_set (G_OBJECT (stream->udpsink[0]), "ttl", ttl, NULL);
+
+    if (stream->udpsrc[1]) {
+      /* configure socket, we give it the same UDP socket as the udpsrc for RTCP
+       * because some servers check the port number of where it sends RTCP to identify
+       * the RTCP packets it receives */
+      g_object_get (G_OBJECT (stream->udpsrc[1]), "used-socket", &socket, NULL);
+      GST_DEBUG_OBJECT (src, "RTCP UDP src has sock %p", socket);
+      /* configure socket and make sure udpsink does not close it when shutting
+       * down, it belongs to udpsrc after all. */
+      g_object_set (G_OBJECT (stream->udpsink[1]), "socket", socket,
+          "close-socket", FALSE, NULL);
+      g_object_unref (socket);
+    }
+
+    /* we don't want to consider this a sink */
+    GST_OBJECT_FLAG_UNSET (stream->udpsink[1], GST_ELEMENT_FLAG_SINK);
+
+    /* we keep this playing always */
+    gst_element_set_locked_state (stream->udpsink[1], TRUE);
+    gst_element_set_state (stream->udpsink[1], GST_STATE_PLAYING);
+
+    gst_object_ref (stream->udpsink[1]);
+    gst_bin_add (GST_BIN_CAST (src), stream->udpsink[1]);
+
+    stream->rtcppad = gst_element_get_static_pad (stream->udpsink[1], "sink");
+
+    /* get session RTCP pad */
+    name = g_strdup_printf ("send_rtcp_src_%u", stream->id);
+    pad = gst_element_get_request_pad (src->manager, name);
+    g_free (name);
+
+    /* and link */
+    if (pad) {
+      gst_pad_link_full (pad, stream->rtcppad, GST_PAD_LINK_CHECK_NOTHING);
+      gst_object_unref (pad);
+    }
+  }
+
+  return TRUE;
+
+  /* ERRORS */
+no_destination:
+  {
+    GST_DEBUG_OBJECT (src, "no destination address specified");
+    return FALSE;
+  }
+no_sink_element:
+  {
+    GST_DEBUG_OBJECT (src, "no UDP sink element found");
+    return FALSE;
+  }
+no_fakesrc_element:
+  {
+    GST_DEBUG_OBJECT (src, "no fakesrc element found");
+    return FALSE;
+  }
+}
+
+/* sets up all elements needed for streaming over the specified transport.
+ * Does not yet expose the element pads, this will be done when there is actuall
+ * dataflow detected, which might never happen when UDP is blocked in a
+ * firewall, for example.
+ */
+static gboolean
+gst_rtspsrc_stream_configure_transport (GstRTSPStream * stream,
+    GstRTSPTransport * transport)
+{
+  GstRTSPSrc *src;
+  GstPad *outpad = NULL;
+  GstPadTemplate *template;
+  gchar *name;
+  const gchar *media_type;
+  guint i, len;
+
+  src = stream->parent;
+
+  GST_DEBUG_OBJECT (src, "configuring transport for stream %p", stream);
+
+  /* get the proper media type for this stream now */
+  if (gst_rtsp_transport_get_media_type (transport, &media_type) < 0)
+    goto unknown_transport;
+  if (!media_type)
+    goto unknown_transport;
+
+  /* configure the final media type */
+  GST_DEBUG_OBJECT (src, "setting media type to %s", media_type);
+
+  len = stream->ptmap->len;
+  for (i = 0; i < len; i++) {
+    GstStructure *s;
+    PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
+
+    if (item->caps == NULL)
+      continue;
+
+    s = gst_caps_get_structure (item->caps, 0);
+    gst_structure_set_name (s, media_type);
+    /* set ssrc if known */
+    if (transport->ssrc)
+      gst_structure_set (s, "ssrc", G_TYPE_UINT, transport->ssrc, NULL);
+  }
+
+  /* try to get and configure a manager, channelpad[0-1] will be configured with
+   * the pads for the manager, or NULL when no manager is needed. */
+  if (!gst_rtspsrc_stream_configure_manager (src, stream, transport))
+    goto no_manager;
+
+  switch (transport->lower_transport) {
+    case GST_RTSP_LOWER_TRANS_TCP:
+      if (!gst_rtspsrc_stream_configure_tcp (src, stream, transport, &outpad))
+        goto transport_failed;
+      break;
+    case GST_RTSP_LOWER_TRANS_UDP_MCAST:
+      if (!gst_rtspsrc_stream_configure_mcast (src, stream, transport, &outpad))
+        goto transport_failed;
+      /* fallthrough, the rest is the same for UDP and MCAST */
+    case GST_RTSP_LOWER_TRANS_UDP:
+      if (!gst_rtspsrc_stream_configure_udp (src, stream, transport, &outpad))
+        goto transport_failed;
+      /* configure udpsinks back to the server for RTCP messages and for the
+       * dummy RTP messages to open NAT. */
+      if (!gst_rtspsrc_stream_configure_udp_sinks (src, stream, transport))
+        goto transport_failed;
+      break;
+    default:
+      goto unknown_transport;
+  }
+
+  if (outpad) {
+    GST_DEBUG_OBJECT (src, "creating ghostpad");
+
+    gst_pad_use_fixed_caps (outpad);
+
+    /* create ghostpad, don't add just yet, this will be done when we activate
+     * the stream. */
+    name = g_strdup_printf ("stream_%u", stream->id);
+    template = gst_static_pad_template_get (&rtptemplate);
+    stream->srcpad = gst_ghost_pad_new_from_template (name, outpad, template);
+    gst_pad_set_event_function (stream->srcpad, gst_rtspsrc_handle_src_event);
+    gst_pad_set_query_function (stream->srcpad, gst_rtspsrc_handle_src_query);
+    gst_object_unref (template);
+    g_free (name);
+
+    gst_object_unref (outpad);
+  }
+  /* mark pad as ok */
+  stream->last_ret = GST_FLOW_OK;
+
+  return TRUE;
+
+  /* ERRORS */
+transport_failed:
+  {
+    GST_DEBUG_OBJECT (src, "failed to configure transport");
+    return FALSE;
+  }
+unknown_transport:
+  {
+    GST_DEBUG_OBJECT (src, "unknown transport");
+    return FALSE;
+  }
+no_manager:
+  {
+    GST_DEBUG_OBJECT (src, "cannot get a session manager");
+    return FALSE;
+  }
+}
+
+/* send a couple of dummy random packets on the receiver RTP port to the server,
+ * this should make a firewall think we initiated the data transfer and
+ * hopefully allow packets to go from the sender port to our RTP receiver port */
+static gboolean
+gst_rtspsrc_send_dummy_packets (GstRTSPSrc * src)
+{
+  GList *walk;
+
+  if (src->nat_method != GST_RTSP_NAT_DUMMY)
+    return TRUE;
+
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+
+    if (stream->fakesrc && stream->udpsink[0]) {
+      GST_DEBUG_OBJECT (src, "sending dummy packet to stream %p", stream);
+      gst_element_set_state (stream->udpsink[0], GST_STATE_NULL);
+      gst_element_set_state (stream->fakesrc, GST_STATE_NULL);
+      gst_element_set_state (stream->udpsink[0], GST_STATE_PLAYING);
+      gst_element_set_state (stream->fakesrc, GST_STATE_PLAYING);
+    }
+  }
+  return TRUE;
+}
+
+/* Adds the source pads of all configured streams to the element.
+ * This code is performed when we detected dataflow.
+ *
+ * We detect dataflow from either the _loop function or with pad probes on the
+ * udp sources.
+ */
+static gboolean
+gst_rtspsrc_activate_streams (GstRTSPSrc * src)
+{
+  GList *walk;
+
+  GST_DEBUG_OBJECT (src, "activating streams");
+
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+
+    if (stream->udpsrc[0]) {
+      /* remove timeout, we are streaming now and timeouts will be handled by
+       * the session manager and jitter buffer */
+      g_object_set (G_OBJECT (stream->udpsrc[0]), "timeout", (guint64) 0, NULL);
+    }
+    if (stream->srcpad) {
+      GST_DEBUG_OBJECT (src, "activating stream pad %p", stream);
+      gst_pad_set_active (stream->srcpad, TRUE);
+
+      /* if we don't have a session manager, set the caps now. If we have a
+       * session, we will get a notification of the pad and the caps. */
+      if (!src->manager) {
+        GstCaps *caps;
+
+        caps = stream_get_caps_for_pt (stream, stream->default_pt);
+        GST_DEBUG_OBJECT (src, "setting pad caps for stream %p", stream);
+        gst_pad_set_caps (stream->srcpad, caps);
+      }
+      /* add the pad */
+      if (!stream->added) {
+        GST_DEBUG_OBJECT (src, "adding stream pad %p", stream);
+        gst_element_add_pad (GST_ELEMENT_CAST (src), stream->srcpad);
+        stream->added = TRUE;
+      }
+    }
+  }
+
+  /* unblock all pads */
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+
+    if (stream->blockid) {
+      GST_DEBUG_OBJECT (src, "unblocking stream pad %p", stream);
+      gst_pad_remove_probe (stream->blockedpad, stream->blockid);
+      stream->blockid = 0;
+    }
+  }
+
+  return TRUE;
+}
+
+static void
+gst_rtspsrc_configure_caps (GstRTSPSrc * src, GstSegment * segment,
+    gboolean reset_manager)
+{
+  GList *walk;
+  guint64 start, stop;
+  gdouble play_speed, play_scale;
+
+  GST_DEBUG_OBJECT (src, "configuring stream caps");
+
+  start = segment->position;
+  stop = segment->duration;
+  play_speed = segment->rate;
+  play_scale = segment->applied_rate;
+
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+    guint j, len;
+
+    if (!stream->setup)
+      continue;
+
+    len = stream->ptmap->len;
+    for (j = 0; j < len; j++) {
+      GstCaps *caps;
+      PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, j);
+
+      if (item->caps == NULL)
+        continue;
+
+      caps = gst_caps_make_writable (item->caps);
+      /* update caps */
+      if (stream->timebase != -1)
+        gst_caps_set_simple (caps, "clock-base", G_TYPE_UINT,
+            (guint) stream->timebase, NULL);
+      if (stream->seqbase != -1)
+        gst_caps_set_simple (caps, "seqnum-base", G_TYPE_UINT,
+            (guint) stream->seqbase, NULL);
+      gst_caps_set_simple (caps, "npt-start", G_TYPE_UINT64, start, NULL);
+      if (stop != -1)
+        gst_caps_set_simple (caps, "npt-stop", G_TYPE_UINT64, stop, NULL);
+      gst_caps_set_simple (caps, "play-speed", G_TYPE_DOUBLE, play_speed, NULL);
+      gst_caps_set_simple (caps, "play-scale", G_TYPE_DOUBLE, play_scale, NULL);
+
+      item->caps = caps;
+      GST_DEBUG_OBJECT (src, "stream %p, pt %d, caps %" GST_PTR_FORMAT, stream,
+          item->pt, caps);
+
+      if (item->pt == stream->default_pt && stream->udpsrc[0]) {
+        g_object_set (stream->udpsrc[0], "caps", caps, NULL);
+      }
+    }
+  }
+  if (reset_manager && src->manager) {
+    GST_DEBUG_OBJECT (src, "clear session");
+    g_signal_emit_by_name (src->manager, "clear-pt-map", NULL);
+  }
+}
+
+static GstFlowReturn
+gst_rtspsrc_combine_flows (GstRTSPSrc * src, GstRTSPStream * stream,
+    GstFlowReturn ret)
+{
+  GList *streams;
+
+  /* store the value */
+  stream->last_ret = ret;
+
+  /* if it's success we can return the value right away */
+  if (ret == GST_FLOW_OK)
+    goto done;
+
+  /* any other error that is not-linked can be returned right
+   * away */
+  if (ret != GST_FLOW_NOT_LINKED)
+    goto done;
+
+  /* only return NOT_LINKED if all other pads returned NOT_LINKED */
+  for (streams = src->streams; streams; streams = g_list_next (streams)) {
+    GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
+
+    ret = ostream->last_ret;
+    /* some other return value (must be SUCCESS but we can return
+     * other values as well) */
+    if (ret != GST_FLOW_NOT_LINKED)
+      goto done;
+  }
+  /* if we get here, all other pads were unlinked and we return
+   * NOT_LINKED then */
+done:
+  return ret;
+}
+
+static gboolean
+gst_rtspsrc_stream_push_event (GstRTSPSrc * src, GstRTSPStream * stream,
+    GstEvent * event)
+{
+  gboolean res = TRUE;
+
+  /* only streams that have a connection to the outside world */
+  if (!stream->setup)
+    goto done;
+
+  if (stream->udpsrc[0]) {
+    gst_event_ref (event);
+    res = gst_element_send_event (stream->udpsrc[0], event);
+  } else if (stream->channelpad[0]) {
+    gst_event_ref (event);
+    if (GST_PAD_IS_SRC (stream->channelpad[0]))
+      res = gst_pad_push_event (stream->channelpad[0], event);
+    else
+      res = gst_pad_send_event (stream->channelpad[0], event);
+  }
+
+  if (stream->udpsrc[1]) {
+    gst_event_ref (event);
+    res &= gst_element_send_event (stream->udpsrc[1], event);
+  } else if (stream->channelpad[1]) {
+    gst_event_ref (event);
+    if (GST_PAD_IS_SRC (stream->channelpad[1]))
+      res &= gst_pad_push_event (stream->channelpad[1], event);
+    else
+      res &= gst_pad_send_event (stream->channelpad[1], event);
+  }
+
+done:
+  gst_event_unref (event);
+
+  return res;
+}
+
+static gboolean
+gst_rtspsrc_push_event (GstRTSPSrc * src, GstEvent * event)
+{
+  GList *streams;
+  gboolean res = TRUE;
+
+  for (streams = src->streams; streams; streams = g_list_next (streams)) {
+    GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
+
+    gst_event_ref (event);
+    res &= gst_rtspsrc_stream_push_event (src, ostream, event);
+  }
+  gst_event_unref (event);
+
+  return res;
+}
+
+static GstRTSPResult
+gst_rtsp_conninfo_connect (GstRTSPSrc * src, GstRTSPConnInfo * info,
+    gboolean async)
+{
+  GstRTSPResult res;
+
+  if (info->connection == NULL) {
+    if (info->url == NULL) {
+      GST_DEBUG_OBJECT (src, "parsing uri (%s)...", info->location);
+      if ((res = gst_rtsp_url_parse (info->location, &info->url)) < 0)
+        goto parse_error;
+    }
+
+    /* create connection */
+    GST_DEBUG_OBJECT (src, "creating connection (%s)...", info->location);
+    if ((res = gst_rtsp_connection_create (info->url, &info->connection)) < 0)
+      goto could_not_create;
+
+    if (info->url_str)
+      g_free (info->url_str);
+    info->url_str = gst_rtsp_url_get_request_uri (info->url);
+
+    GST_DEBUG_OBJECT (src, "sanitized uri %s", info->url_str);
+
+    if (info->url->transports & GST_RTSP_LOWER_TRANS_TLS) {
+      if (!gst_rtsp_connection_set_tls_validation_flags (info->connection,
+              src->tls_validation_flags))
+        GST_WARNING_OBJECT (src, "Unable to set TLS validation flags");
+
+      if (src->tls_database)
+        gst_rtsp_connection_set_tls_database (info->connection,
+            src->tls_database);
+    }
+
+    if (info->url->transports & GST_RTSP_LOWER_TRANS_HTTP)
+      gst_rtsp_connection_set_tunneled (info->connection, TRUE);
+
+    if (src->proxy_host) {
+      GST_DEBUG_OBJECT (src, "setting proxy %s:%d", src->proxy_host,
+          src->proxy_port);
+      gst_rtsp_connection_set_proxy (info->connection, src->proxy_host,
+          src->proxy_port);
+    }
+  }
+
+  if (!info->connected) {
+    /* connect */
+    if (async)
+      GST_ELEMENT_PROGRESS (src, CONTINUE, "connect",
+          ("Connecting to %s", info->location));
+    GST_DEBUG_OBJECT (src, "connecting (%s)...", info->location);
+    if ((res =
+            gst_rtsp_connection_connect (info->connection,
+                src->ptcp_timeout)) < 0)
+      goto could_not_connect;
+
+    info->connected = TRUE;
+  }
+  return GST_RTSP_OK;
+
+  /* ERRORS */
+parse_error:
+  {
+    GST_ERROR_OBJECT (src, "No valid RTSP URL was provided");
+    return res;
+  }
+could_not_create:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+    GST_ERROR_OBJECT (src, "Could not create connection. (%s)", str);
+    g_free (str);
+    return res;
+  }
+could_not_connect:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+    GST_ERROR_OBJECT (src, "Could not connect to server. (%s)", str);
+    g_free (str);
+    return res;
+  }
+}
+
+static GstRTSPResult
+gst_rtsp_conninfo_close (GstRTSPSrc * src, GstRTSPConnInfo * info,
+    gboolean free)
+{
+  GST_RTSP_STATE_LOCK (src);
+  if (info->connected) {
+    GST_DEBUG_OBJECT (src, "closing connection...");
+    gst_rtsp_connection_close (info->connection);
+    info->connected = FALSE;
+  }
+  if (free && info->connection) {
+    /* free connection */
+    GST_DEBUG_OBJECT (src, "freeing connection...");
+    gst_rtsp_connection_free (info->connection);
+    info->connection = NULL;
+  }
+  GST_RTSP_STATE_UNLOCK (src);
+  return GST_RTSP_OK;
+}
+
+static GstRTSPResult
+gst_rtsp_conninfo_reconnect (GstRTSPSrc * src, GstRTSPConnInfo * info,
+    gboolean async)
+{
+  GstRTSPResult res;
+
+  GST_DEBUG_OBJECT (src, "reconnecting connection...");
+  gst_rtsp_conninfo_close (src, info, FALSE);
+  res = gst_rtsp_conninfo_connect (src, info, async);
+
+  return res;
+}
+
+static void
+gst_rtspsrc_connection_flush (GstRTSPSrc * src, gboolean flush)
+{
+  GList *walk;
+
+  GST_DEBUG_OBJECT (src, "set flushing %d", flush);
+  GST_RTSP_STATE_LOCK (src);
+  if (src->conninfo.connection && src->conninfo.flushing != flush) {
+    GST_DEBUG_OBJECT (src, "connection flush");
+    gst_rtsp_connection_flush (src->conninfo.connection, flush);
+    src->conninfo.flushing = flush;
+  }
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+    if (stream->conninfo.connection && stream->conninfo.flushing != flush) {
+      GST_DEBUG_OBJECT (src, "stream %p flush", stream);
+      gst_rtsp_connection_flush (stream->conninfo.connection, flush);
+      stream->conninfo.flushing = flush;
+    }
+  }
+  GST_RTSP_STATE_UNLOCK (src);
+}
+
+/* FIXME, handle server request, reply with OK, for now */
+static GstRTSPResult
+gst_rtspsrc_handle_request (GstRTSPSrc * src, GstRTSPConnection * conn,
+    GstRTSPMessage * request)
+{
+  GstRTSPMessage response = { 0 };
+  GstRTSPResult res;
+
+  GST_DEBUG_OBJECT (src, "got server request message");
+
+  if (src->debug)
+    gst_rtsp_message_dump (request);
+
+  res = gst_rtsp_ext_list_receive_request (src->extensions, request);
+
+  if (res == GST_RTSP_ENOTIMPL) {
+    /* default implementation, send OK */
+    GST_DEBUG_OBJECT (src, "prepare OK reply");
+    res =
+        gst_rtsp_message_init_response (&response, GST_RTSP_STS_OK, "OK",
+        request);
+    if (res < 0)
+      goto send_error;
+
+    /* let app parse and reply */
+    g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_HANDLE_REQUEST],
+        0, request, &response);
+
+    if (src->debug)
+      gst_rtsp_message_dump (&response);
+
+    res = gst_rtspsrc_connection_send (src, conn, &response, NULL);
+    if (res < 0)
+      goto send_error;
+
+    gst_rtsp_message_unset (&response);
+  } else if (res == GST_RTSP_EEOF)
+    return res;
+
+  return GST_RTSP_OK;
+
+  /* ERRORS */
+send_error:
+  {
+    gst_rtsp_message_unset (&response);
+    return res;
+  }
+}
+
+/* send server keep-alive */
+static GstRTSPResult
+gst_rtspsrc_send_keep_alive (GstRTSPSrc * src)
+{
+  GstRTSPMessage request = { 0 };
+  GstRTSPResult res;
+  GstRTSPMethod method;
+  const gchar *control;
+
+  if (src->do_rtsp_keep_alive == FALSE) {
+    GST_DEBUG_OBJECT (src, "do-rtsp-keep-alive is FALSE, not sending.");
+    gst_rtsp_connection_reset_timeout (src->conninfo.connection);
+    return GST_RTSP_OK;
+  }
+
+  GST_DEBUG_OBJECT (src, "creating server keep-alive");
+
+  /* find a method to use for keep-alive */
+  if (src->methods & GST_RTSP_GET_PARAMETER)
+    method = GST_RTSP_GET_PARAMETER;
+  else
+    method = GST_RTSP_OPTIONS;
+
+  control = get_aggregate_control (src);
+  if (control == NULL)
+    goto no_control;
+
+  res = gst_rtsp_message_init_request (&request, method, control);
+  if (res < 0)
+    goto send_error;
+
+  if (src->debug)
+    gst_rtsp_message_dump (&request);
+
+  res =
+      gst_rtspsrc_connection_send (src, src->conninfo.connection, &request,
+      NULL);
+  if (res < 0)
+    goto send_error;
+
+  gst_rtsp_connection_reset_timeout (src->conninfo.connection);
+  gst_rtsp_message_unset (&request);
+
+  return GST_RTSP_OK;
+
+  /* ERRORS */
+no_control:
+  {
+    GST_WARNING_OBJECT (src, "no control url to send keepalive");
+    return GST_RTSP_OK;
+  }
+send_error:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    gst_rtsp_message_unset (&request);
+    GST_ELEMENT_WARNING (src, RESOURCE, WRITE, (NULL),
+        ("Could not send keep-alive. (%s)", str));
+    g_free (str);
+    return res;
+  }
+}
+
+static GstFlowReturn
+gst_rtspsrc_handle_data (GstRTSPSrc * src, GstRTSPMessage * message)
+{
+  GstFlowReturn ret = GST_FLOW_OK;
+  gint channel;
+  GstRTSPStream *stream;
+  GstPad *outpad = NULL;
+  guint8 *data;
+  guint size;
+  GstBuffer *buf;
+  gboolean is_rtcp;
+
+  channel = message->type_data.data.channel;
+
+  stream = find_stream (src, &channel, (gpointer) find_stream_by_channel);
+  if (!stream)
+    goto unknown_stream;
+
+  if (channel == stream->channel[0]) {
+    outpad = stream->channelpad[0];
+    is_rtcp = FALSE;
+  } else if (channel == stream->channel[1]) {
+    outpad = stream->channelpad[1];
+    is_rtcp = TRUE;
+  } else {
+    is_rtcp = FALSE;
+  }
+
+  /* take a look at the body to figure out what we have */
+  gst_rtsp_message_get_body (message, &data, &size);
+  if (size < 2)
+    goto invalid_length;
+
+  /* channels are not correct on some servers, do extra check */
+  if (data[1] >= 200 && data[1] <= 204) {
+    /* hmm RTCP message switch to the RTCP pad of the same stream. */
+    outpad = stream->channelpad[1];
+    is_rtcp = TRUE;
+  }
+
+  /* we have no clue what this is, just ignore then. */
+  if (outpad == NULL)
+    goto unknown_stream;
+
+  /* take the message body for further processing */
+  gst_rtsp_message_steal_body (message, &data, &size);
+
+  /* strip the trailing \0 */
+  size -= 1;
+
+  buf = gst_buffer_new ();
+  gst_buffer_append_memory (buf,
+      gst_memory_new_wrapped (0, data, size, 0, size, data, g_free));
+
+  /* don't need message anymore */
+  gst_rtsp_message_unset (message);
+
+  GST_DEBUG_OBJECT (src, "pushing data of size %d on channel %d", size,
+      channel);
+
+  if (src->need_activate) {
+    gchar *stream_id;
+    GstEvent *event;
+    GChecksum *cs;
+    gchar *uri;
+    GList *streams;
+    guint group_id = gst_util_group_id_next ();
+
+    /* generate an SHA256 sum of the URI */
+    cs = g_checksum_new (G_CHECKSUM_SHA256);
+    uri = src->conninfo.location;
+    g_checksum_update (cs, (const guchar *) uri, strlen (uri));
+
+    for (streams = src->streams; streams; streams = g_list_next (streams)) {
+      GstRTSPStream *ostream = (GstRTSPStream *) streams->data;
+      GstCaps *caps;
+
+      stream_id =
+          g_strdup_printf ("%s/%d", g_checksum_get_string (cs), ostream->id);
+      event = gst_event_new_stream_start (stream_id);
+      gst_event_set_group_id (event, group_id);
+
+      g_free (stream_id);
+      gst_rtspsrc_stream_push_event (src, ostream, event);
+
+      if ((caps = stream_get_caps_for_pt (ostream, ostream->default_pt))) {
+        /* only streams that have a connection to the outside world */
+        if (ostream->setup) {
+          if (ostream->udpsrc[0]) {
+            gst_element_send_event (ostream->udpsrc[0],
+                gst_event_new_caps (caps));
+          } else if (ostream->channelpad[0]) {
+            if (GST_PAD_IS_SRC (ostream->channelpad[0]))
+              gst_pad_push_event (ostream->channelpad[0],
+                  gst_event_new_caps (caps));
+            else
+              gst_pad_send_event (ostream->channelpad[0],
+                  gst_event_new_caps (caps));
+          }
+
+          caps = gst_caps_new_empty_simple ("application/x-rtcp");
+
+          if (ostream->udpsrc[1]) {
+            gst_element_send_event (ostream->udpsrc[1],
+                gst_event_new_caps (caps));
+          } else if (ostream->channelpad[1]) {
+            if (GST_PAD_IS_SRC (ostream->channelpad[1]))
+              gst_pad_push_event (ostream->channelpad[1],
+                  gst_event_new_caps (caps));
+            else
+              gst_pad_send_event (ostream->channelpad[1],
+                  gst_event_new_caps (caps));
+          }
+
+          gst_caps_unref (caps);
+        }
+      }
+    }
+    g_checksum_free (cs);
+
+    gst_rtspsrc_activate_streams (src);
+    src->need_activate = FALSE;
+    src->need_segment = TRUE;
+  }
+
+  if (src->base_time == -1) {
+    /* Take current running_time. This timestamp will be put on
+     * the first buffer of each stream because we are a live source and so we
+     * timestamp with the running_time. When we are dealing with TCP, we also
+     * only timestamp the first buffer (using the DISCONT flag) because a server
+     * typically bursts data, for which we don't want to compensate by speeding
+     * up the media. The other timestamps will be interpollated from this one
+     * using the RTP timestamps. */
+    GST_OBJECT_LOCK (src);
+    if (GST_ELEMENT_CLOCK (src)) {
+      GstClockTime now;
+      GstClockTime base_time;
+
+      now = gst_clock_get_time (GST_ELEMENT_CLOCK (src));
+      base_time = GST_ELEMENT_CAST (src)->base_time;
+
+      src->base_time = now - base_time;
+
+      GST_DEBUG_OBJECT (src, "first buffer at time %" GST_TIME_FORMAT ", base %"
+          GST_TIME_FORMAT, GST_TIME_ARGS (now), GST_TIME_ARGS (base_time));
+    }
+    GST_OBJECT_UNLOCK (src);
+  }
+
+  /* If needed send a new segment, don't forget we are live and buffer are
+   * timestamped with running time */
+  if (src->need_segment) {
+    GstSegment segment;
+    src->need_segment = FALSE;
+    gst_segment_init (&segment, GST_FORMAT_TIME);
+    gst_rtspsrc_push_event (src, gst_event_new_segment (&segment));
+  }
+
+  if (stream->discont && !is_rtcp) {
+    /* mark first RTP buffer as discont */
+    GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
+    stream->discont = FALSE;
+    /* first buffer gets the timestamp, other buffers are not timestamped and
+     * their presentation time will be interpollated from the rtp timestamps. */
+    GST_DEBUG_OBJECT (src, "setting timestamp %" GST_TIME_FORMAT,
+        GST_TIME_ARGS (src->base_time));
+
+    GST_BUFFER_TIMESTAMP (buf) = src->base_time;
+  }
+
+  /* chain to the peer pad */
+  if (GST_PAD_IS_SINK (outpad))
+    ret = gst_pad_chain (outpad, buf);
+  else
+    ret = gst_pad_push (outpad, buf);
+
+  if (!is_rtcp) {
+    /* combine all stream flows for the data transport */
+    ret = gst_rtspsrc_combine_flows (src, stream, ret);
+  }
+  return ret;
+
+  /* ERRORS */
+unknown_stream:
+  {
+    GST_DEBUG_OBJECT (src, "unknown stream on channel %d, ignored", channel);
+    gst_rtsp_message_unset (message);
+    return GST_FLOW_OK;
+  }
+invalid_length:
+  {
+    GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
+        ("Short message received, ignoring."));
+    gst_rtsp_message_unset (message);
+    return GST_FLOW_OK;
+  }
+}
+
+static GstFlowReturn
+gst_rtspsrc_loop_interleaved (GstRTSPSrc * src)
+{
+  GstRTSPMessage message = { 0 };
+  GstRTSPResult res;
+  GstFlowReturn ret = GST_FLOW_OK;
+  GTimeVal tv_timeout;
+
+  while (TRUE) {
+    /* get the next timeout interval */
+    gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
+
+    /* see if the timeout period expired */
+    if ((tv_timeout.tv_sec | tv_timeout.tv_usec) == 0) {
+      GST_DEBUG_OBJECT (src, "timout, sending keep-alive");
+      /* send keep-alive, only act on interrupt, a warning will be posted for
+       * other errors. */
+      if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
+        goto interrupt;
+      /* get new timeout */
+      gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
+    }
+
+    GST_DEBUG_OBJECT (src, "doing receive with timeout %ld seconds, %ld usec",
+        tv_timeout.tv_sec, tv_timeout.tv_usec);
+
+    /* protect the connection with the connection lock so that we can see when
+     * we are finished doing server communication */
+    res =
+        gst_rtspsrc_connection_receive (src, src->conninfo.connection,
+        &message, src->ptcp_timeout);
+
+    switch (res) {
+      case GST_RTSP_OK:
+        GST_DEBUG_OBJECT (src, "we received a server message");
+        break;
+      case GST_RTSP_EINTR:
+        /* we got interrupted this means we need to stop */
+        goto interrupt;
+      case GST_RTSP_ETIMEOUT:
+        /* no reply, send keep alive */
+        GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
+        if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
+          goto interrupt;
+        continue;
+      case GST_RTSP_EEOF:
+        /* go EOS when the server closed the connection */
+        goto server_eof;
+      default:
+        goto receive_error;
+    }
+
+    switch (message.type) {
+      case GST_RTSP_MESSAGE_REQUEST:
+        /* server sends us a request message, handle it */
+        res =
+            gst_rtspsrc_handle_request (src, src->conninfo.connection,
+            &message);
+        if (res == GST_RTSP_EEOF)
+          goto server_eof;
+        else if (res < 0)
+          goto handle_request_failed;
+        break;
+      case GST_RTSP_MESSAGE_RESPONSE:
+        /* we ignore response messages */
+        GST_DEBUG_OBJECT (src, "ignoring response message");
+        if (src->debug)
+          gst_rtsp_message_dump (&message);
+        break;
+      case GST_RTSP_MESSAGE_DATA:
+        GST_DEBUG_OBJECT (src, "got data message");
+        ret = gst_rtspsrc_handle_data (src, &message);
+        if (ret != GST_FLOW_OK)
+          goto handle_data_failed;
+        break;
+      default:
+        GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
+            message.type);
+        break;
+    }
+  }
+  g_assert_not_reached ();
+
+  /* ERRORS */
+server_eof:
+  {
+    GST_DEBUG_OBJECT (src, "we got an eof from the server");
+    GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
+        ("The server closed the connection."));
+    src->conninfo.connected = FALSE;
+    gst_rtsp_message_unset (&message);
+    return GST_FLOW_EOS;
+  }
+interrupt:
+  {
+    gst_rtsp_message_unset (&message);
+    GST_DEBUG_OBJECT (src, "got interrupted");
+    return GST_FLOW_FLUSHING;
+  }
+receive_error:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
+        ("Could not receive message. (%s)", str));
+    g_free (str);
+
+    gst_rtsp_message_unset (&message);
+    return GST_FLOW_ERROR;
+  }
+handle_request_failed:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
+        ("Could not handle server message. (%s)", str));
+    g_free (str);
+    gst_rtsp_message_unset (&message);
+    return GST_FLOW_ERROR;
+  }
+handle_data_failed:
+  {
+    GST_DEBUG_OBJECT (src, "could no handle data message");
+    return ret;
+  }
+}
+
+static GstFlowReturn
+gst_rtspsrc_loop_udp (GstRTSPSrc * src)
+{
+  GstRTSPResult res;
+  GstRTSPMessage message = { 0 };
+  gint retry = 0;
+
+  while (TRUE) {
+    GTimeVal tv_timeout;
+
+    /* get the next timeout interval */
+    gst_rtsp_connection_next_timeout (src->conninfo.connection, &tv_timeout);
+
+    GST_DEBUG_OBJECT (src, "doing receive with timeout %d seconds",
+        (gint) tv_timeout.tv_sec);
+
+    gst_rtsp_message_unset (&message);
+
+    /* we should continue reading the TCP socket because the server might
+     * send us requests. When the session timeout expires, we need to send a
+     * keep-alive request to keep the session open. */
+    res = gst_rtspsrc_connection_receive (src, src->conninfo.connection,
+        &message, &tv_timeout);
+
+    switch (res) {
+      case GST_RTSP_OK:
+        GST_DEBUG_OBJECT (src, "we received a server message");
+        break;
+      case GST_RTSP_EINTR:
+        /* we got interrupted, see what we have to do */
+        goto interrupt;
+      case GST_RTSP_ETIMEOUT:
+        /* send keep-alive, ignore the result, a warning will be posted. */
+        GST_DEBUG_OBJECT (src, "timeout, sending keep-alive");
+        if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
+          goto interrupt;
+        continue;
+      case GST_RTSP_EEOF:
+        /* server closed the connection. not very fatal for UDP, reconnect and
+         * see what happens. */
+        GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
+            ("The server closed the connection."));
+        if (src->udp_reconnect) {
+          if ((res =
+                  gst_rtsp_conninfo_reconnect (src, &src->conninfo, FALSE)) < 0)
+            goto connect_error;
+        } else {
+          goto server_eof;
+        }
+        continue;
+      case GST_RTSP_ENET:
+        GST_DEBUG_OBJECT (src, "An ethernet problem occured.");
+      default:
+        GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
+            ("Unhandled return value %d.", res));
+        goto receive_error;
+    }
+
+    switch (message.type) {
+      case GST_RTSP_MESSAGE_REQUEST:
+        /* server sends us a request message, handle it */
+        res =
+            gst_rtspsrc_handle_request (src, src->conninfo.connection,
+            &message);
+        if (res == GST_RTSP_EEOF)
+          goto server_eof;
+        else if (res < 0)
+          goto handle_request_failed;
+        break;
+      case GST_RTSP_MESSAGE_RESPONSE:
+        /* we ignore response and data messages */
+        GST_DEBUG_OBJECT (src, "ignoring response message");
+        if (src->debug)
+          gst_rtsp_message_dump (&message);
+        if (message.type_data.response.code == GST_RTSP_STS_UNAUTHORIZED) {
+          GST_DEBUG_OBJECT (src, "but is Unauthorized response ...");
+          if (gst_rtspsrc_setup_auth (src, &message) && !(retry++)) {
+            GST_DEBUG_OBJECT (src, "so retrying keep-alive");
+            if ((res = gst_rtspsrc_send_keep_alive (src)) == GST_RTSP_EINTR)
+              goto interrupt;
+          }
+        } else {
+          retry = 0;
+        }
+        break;
+      case GST_RTSP_MESSAGE_DATA:
+        /* we ignore response and data messages */
+        GST_DEBUG_OBJECT (src, "ignoring data message");
+        break;
+      default:
+        GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
+            message.type);
+        break;
+    }
+  }
+  g_assert_not_reached ();
+
+  /* we get here when the connection got interrupted */
+interrupt:
+  {
+    gst_rtsp_message_unset (&message);
+    GST_DEBUG_OBJECT (src, "got interrupted");
+    return GST_FLOW_FLUSHING;
+  }
+connect_error:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+    GstFlowReturn ret;
+
+    src->conninfo.connected = FALSE;
+    if (res != GST_RTSP_EINTR) {
+      GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
+          ("Could not connect to server. (%s)", str));
+      g_free (str);
+      ret = GST_FLOW_ERROR;
+    } else {
+      ret = GST_FLOW_FLUSHING;
+    }
+    return ret;
+  }
+receive_error:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
+        ("Could not receive message. (%s)", str));
+    g_free (str);
+    return GST_FLOW_ERROR;
+  }
+handle_request_failed:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+    GstFlowReturn ret;
+
+    gst_rtsp_message_unset (&message);
+    if (res != GST_RTSP_EINTR) {
+      GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
+          ("Could not handle server message. (%s)", str));
+      g_free (str);
+      ret = GST_FLOW_ERROR;
+    } else {
+      ret = GST_FLOW_FLUSHING;
+    }
+    return ret;
+  }
+server_eof:
+  {
+    GST_DEBUG_OBJECT (src, "we got an eof from the server");
+    GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
+        ("The server closed the connection."));
+    src->conninfo.connected = FALSE;
+    gst_rtsp_message_unset (&message);
+    return GST_FLOW_EOS;
+  }
+}
+
+static GstRTSPResult
+gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  gboolean restart;
+
+  GST_DEBUG_OBJECT (src, "doing reconnect");
+
+  GST_OBJECT_LOCK (src);
+  /* only restart when the pads were not yet activated, else we were
+   * streaming over UDP */
+  restart = src->need_activate;
+  GST_OBJECT_UNLOCK (src);
+
+  /* no need to restart, we're done */
+  if (!restart)
+    goto done;
+
+  /* we can try only TCP now */
+  src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
+
+  /* close and cleanup our state */
+  if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
+    goto done;
+
+  /* see if we have TCP left to try. Also don't try TCP when we were configured
+   * with an SDP. */
+  if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
+    goto no_protocols;
+
+  /* We post a warning message now to inform the user
+   * that nothing happened. It's most likely a firewall thing. */
+  GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
+      ("Could not receive any UDP packets for %.4f seconds, maybe your "
+          "firewall is blocking it. Retrying using a TCP connection.",
+          gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
+
+  /* open new connection using tcp */
+  if (gst_rtspsrc_open (src, async) < 0)
+    goto open_failed;
+
+  /* start playback */
+  if (gst_rtspsrc_play (src, &src->segment, async) < 0)
+    goto play_failed;
+
+done:
+  return res;
+
+  /* ERRORS */
+no_protocols:
+  {
+    src->cur_protocols = 0;
+    /* no transport possible, post an error and stop */
+    GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
+        ("Could not receive any UDP packets for %.4f seconds, maybe your "
+            "firewall is blocking it. No other protocols to try.",
+            gst_guint64_to_gdouble (src->udp_timeout / 1000000.0)));
+    return GST_RTSP_ERROR;
+  }
+open_failed:
+  {
+    GST_DEBUG_OBJECT (src, "open failed");
+    return GST_RTSP_OK;
+  }
+play_failed:
+  {
+    GST_DEBUG_OBJECT (src, "play failed");
+    return GST_RTSP_OK;
+  }
+}
+
+static void
+gst_rtspsrc_loop_start_cmd (GstRTSPSrc * src, gint cmd)
+{
+  switch (cmd) {
+    case CMD_OPEN:
+      GST_ELEMENT_PROGRESS (src, START, "open", ("Opening Stream"));
+      break;
+    case CMD_PLAY:
+      GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PLAY request"));
+      break;
+    case CMD_PAUSE:
+      GST_ELEMENT_PROGRESS (src, START, "request", ("Sending PAUSE request"));
+      break;
+    case CMD_CLOSE:
+      GST_ELEMENT_PROGRESS (src, START, "close", ("Closing Stream"));
+      break;
+    default:
+      break;
+  }
+}
+
+static void
+gst_rtspsrc_loop_complete_cmd (GstRTSPSrc * src, gint cmd)
+{
+
+#ifdef GST_EXT_RTSP_MODIFICATION
+  GstStructure *s;
+#endif
+
+  switch (cmd) {
+    case CMD_OPEN:
+#ifdef GST_EXT_RTSP_MODIFICATION
+      s = gst_structure_new ("RTSP_duration", "duration", G_TYPE_UINT64, src->segment.duration, NULL);
+      gst_element_post_message (GST_ELEMENT_CAST (src), gst_message_new_custom(GST_MESSAGE_ELEMENT, GST_OBJECT(src), s));
+      GST_DEBUG_OBJECT(src, "GST_MESSAGE_ELEMENT msg posting. duration=%"GST_TIME_FORMAT, GST_TIME_ARGS(src->segment.duration));
+#endif
+      GST_ELEMENT_PROGRESS (src, COMPLETE, "open", ("Opened Stream"));
+#ifdef GST_EXT_RTSP_MODIFICATION
+      /* rtspsrc PAUSE state should be here for parsing sdp before PAUSE state changed. */
+      g_mutex_lock(&(src)->pause_lock);
+      g_cond_signal (&(src)->open_end);
+      g_mutex_unlock(&(src)->pause_lock);
+#endif
+      break;
+    case CMD_PLAY:
+      GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PLAY request"));
+      break;
+    case CMD_PAUSE:
+      GST_ELEMENT_PROGRESS (src, COMPLETE, "request", ("Sent PAUSE request"));
+      break;
+    case CMD_CLOSE:
+      GST_ELEMENT_PROGRESS (src, COMPLETE, "close", ("Closed Stream"));
+      break;
+    default:
+      break;
+  }
+}
+
+static void
+gst_rtspsrc_loop_cancel_cmd (GstRTSPSrc * src, gint cmd)
+{
+  switch (cmd) {
+    case CMD_OPEN:
+      GST_ELEMENT_PROGRESS (src, CANCELED, "open", ("Open canceled"));
+      break;
+    case CMD_PLAY:
+      GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PLAY canceled"));
+      break;
+    case CMD_PAUSE:
+      GST_ELEMENT_PROGRESS (src, CANCELED, "request", ("PAUSE canceled"));
+      break;
+    case CMD_CLOSE:
+      GST_ELEMENT_PROGRESS (src, CANCELED, "close", ("Close canceled"));
+      break;
+    default:
+      break;
+  }
+}
+
+static void
+gst_rtspsrc_loop_error_cmd (GstRTSPSrc * src, gint cmd)
+{
+  switch (cmd) {
+    case CMD_OPEN:
+      GST_ELEMENT_PROGRESS (src, ERROR, "open", ("Open failed"));
+      break;
+    case CMD_PLAY:
+      GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PLAY failed"));
+      break;
+    case CMD_PAUSE:
+      GST_ELEMENT_PROGRESS (src, ERROR, "request", ("PAUSE failed"));
+      break;
+    case CMD_CLOSE:
+      GST_ELEMENT_PROGRESS (src, ERROR, "close", ("Close failed"));
+      break;
+    default:
+      break;
+  }
+}
+
+static void
+gst_rtspsrc_loop_end_cmd (GstRTSPSrc * src, gint cmd, GstRTSPResult ret)
+{
+  if (ret == GST_RTSP_OK)
+    gst_rtspsrc_loop_complete_cmd (src, cmd);
+  else if (ret == GST_RTSP_EINTR)
+    gst_rtspsrc_loop_cancel_cmd (src, cmd);
+  else
+    gst_rtspsrc_loop_error_cmd (src, cmd);
+}
+
+static gboolean
+gst_rtspsrc_loop_send_cmd (GstRTSPSrc * src, gint cmd, gint mask)
+{
+  gint old;
+  gboolean flushed = FALSE;
+
+  /* start new request */
+  gst_rtspsrc_loop_start_cmd (src, cmd);
+
+  GST_DEBUG_OBJECT (src, "sending cmd %s", cmd_to_string (cmd));
+
+  GST_OBJECT_LOCK (src);
+  old = src->pending_cmd;
+  if (old == CMD_RECONNECT) {
+    GST_DEBUG_OBJECT (src, "ignore, we were reconnecting");
+    cmd = CMD_RECONNECT;
+  }
+  if (old != CMD_WAIT) {
+    src->pending_cmd = CMD_WAIT;
+    GST_OBJECT_UNLOCK (src);
+    /* cancel previous request */
+    GST_DEBUG_OBJECT (src, "cancel previous request %s", cmd_to_string (old));
+    gst_rtspsrc_loop_cancel_cmd (src, old);
+    GST_OBJECT_LOCK (src);
+  }
+  src->pending_cmd = cmd;
+  /* interrupt if allowed */
+  if (src->busy_cmd & mask) {
+    GST_DEBUG_OBJECT (src, "connection flush busy %s",
+        cmd_to_string (src->busy_cmd));
+    gst_rtspsrc_connection_flush (src, TRUE);
+    flushed = TRUE;
+  } else {
+    GST_DEBUG_OBJECT (src, "not interrupting busy cmd %s",
+        cmd_to_string (src->busy_cmd));
+  }
+  if (src->task)
+    gst_task_start (src->task);
+  GST_OBJECT_UNLOCK (src);
+
+  return flushed;
+}
+
+static gboolean
+gst_rtspsrc_loop (GstRTSPSrc * src)
+{
+  GstFlowReturn ret;
+
+  if (!src->conninfo.connection || !src->conninfo.connected)
+    goto no_connection;
+
+  if (src->interleaved)
+    ret = gst_rtspsrc_loop_interleaved (src);
+  else
+    ret = gst_rtspsrc_loop_udp (src);
+
+  if (ret != GST_FLOW_OK)
+    goto pause;
+
+  return TRUE;
+
+  /* ERRORS */
+no_connection:
+  {
+    GST_WARNING_OBJECT (src, "we are not connected");
+    ret = GST_FLOW_FLUSHING;
+    goto pause;
+  }
+pause:
+  {
+    const gchar *reason = gst_flow_get_name (ret);
+
+    GST_DEBUG_OBJECT (src, "pausing task, reason %s", reason);
+    src->running = FALSE;
+    if (ret == GST_FLOW_EOS) {
+      /* perform EOS logic */
+      if (src->segment.flags & GST_SEEK_FLAG_SEGMENT) {
+        gst_element_post_message (GST_ELEMENT_CAST (src),
+            gst_message_new_segment_done (GST_OBJECT_CAST (src),
+                src->segment.format, src->segment.position));
+        gst_rtspsrc_push_event (src,
+            gst_event_new_segment_done (src->segment.format,
+                src->segment.position));
+      } else {
+        gst_rtspsrc_push_event (src, gst_event_new_eos ());
+      }
+    } else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
+      /* for fatal errors we post an error message, post the error before the
+       * EOS so the app knows about the error first. */
+      GST_ELEMENT_ERROR (src, STREAM, FAILED,
+          ("Internal data flow error."),
+          ("streaming task paused, reason %s (%d)", reason, ret));
+      gst_rtspsrc_push_event (src, gst_event_new_eos ());
+    }
+    gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_LOOP);
+    return FALSE;
+  }
+}
+
+#ifndef GST_DISABLE_GST_DEBUG
+static const gchar *
+gst_rtsp_auth_method_to_string (GstRTSPAuthMethod method)
+{
+  gint index = 0;
+
+  while (method != 0) {
+    index++;
+    method >>= 1;
+  }
+  switch (index) {
+    case 0:
+      return "None";
+    case 1:
+      return "Basic";
+    case 2:
+      return "Digest";
+  }
+
+  return "Unknown";
+}
+#endif
+
+static const gchar *
+gst_rtspsrc_skip_lws (const gchar * s)
+{
+  while (g_ascii_isspace (*s))
+    s++;
+  return s;
+}
+
+static const gchar *
+gst_rtspsrc_unskip_lws (const gchar * s, const gchar * start)
+{
+  while (s > start && g_ascii_isspace (*(s - 1)))
+    s--;
+  return s;
+}
+
+static const gchar *
+gst_rtspsrc_skip_commas (const gchar * s)
+{
+  /* The grammar allows for multiple commas */
+  while (g_ascii_isspace (*s) || *s == ',')
+    s++;
+  return s;
+}
+
+static const gchar *
+gst_rtspsrc_skip_item (const gchar * s)
+{
+  gboolean quoted = FALSE;
+  const gchar *start = s;
+
+  /* A list item ends at the last non-whitespace character
+   * before a comma which is not inside a quoted-string. Or at
+   * the end of the string.
+   */
+  while (*s) {
+    if (*s == '"')
+      quoted = !quoted;
+    else if (quoted) {
+      if (*s == '\\' && *(s + 1))
+        s++;
+    } else {
+      if (*s == ',')
+        break;
+    }
+    s++;
+  }
+
+  return gst_rtspsrc_unskip_lws (s, start);
+}
+
+static void
+gst_rtsp_decode_quoted_string (gchar * quoted_string)
+{
+  gchar *src, *dst;
+
+  src = quoted_string + 1;
+  dst = quoted_string;
+  while (*src && *src != '"') {
+    if (*src == '\\' && *(src + 1))
+      src++;
+    *dst++ = *src++;
+  }
+  *dst = '\0';
+}
+
+/* Extract the authentication tokens that the server provided for each method
+ * into an array of structures and give those to the connection object.
+ */
+static void
+gst_rtspsrc_parse_digest_challenge (GstRTSPConnection * conn,
+    const gchar * header, gboolean * stale)
+{
+  GSList *list = NULL, *iter;
+  const gchar *end;
+  gchar *item, *eq, *name_end, *value;
+
+  g_return_if_fail (stale != NULL);
+
+  gst_rtsp_connection_clear_auth_params (conn);
+  *stale = FALSE;
+
+  /* Parse a header whose content is described by RFC2616 as
+   * "#something", where "something" does not itself contain commas,
+   * except as part of quoted-strings, into a list of allocated strings.
+   */
+  header = gst_rtspsrc_skip_commas (header);
+  while (*header) {
+    end = gst_rtspsrc_skip_item (header);
+    list = g_slist_prepend (list, g_strndup (header, end - header));
+    header = gst_rtspsrc_skip_commas (end);
+  }
+  if (!list)
+    return;
+
+  list = g_slist_reverse (list);
+  for (iter = list; iter; iter = iter->next) {
+    item = iter->data;
+
+    eq = strchr (item, '=');
+    if (eq) {
+      name_end = (gchar *) gst_rtspsrc_unskip_lws (eq, item);
+      if (name_end == item) {
+        /* That's no good... */
+        g_free (item);
+        continue;
+      }
+
+      *name_end = '\0';
+
+      value = (gchar *) gst_rtspsrc_skip_lws (eq + 1);
+      if (*value == '"')
+        gst_rtsp_decode_quoted_string (value);
+    } else
+      value = NULL;
+
+    if (value && strcmp (item, "stale") == 0 && strcmp (value, "TRUE") == 0)
+      *stale = TRUE;
+    gst_rtsp_connection_set_auth_param (conn, item, value);
+    g_free (item);
+  }
+
+  g_slist_free (list);
+}
+
+/* Parse a WWW-Authenticate Response header and determine the
+ * available authentication methods
+ *
+ * This code should also cope with the fact that each WWW-Authenticate
+ * header can contain multiple challenge methods + tokens
+ *
+ * At the moment, for Basic auth, we just do a minimal check and don't
+ * even parse out the realm */
+static void
+gst_rtspsrc_parse_auth_hdr (gchar * hdr, GstRTSPAuthMethod * methods,
+    GstRTSPConnection * conn, gboolean * stale)
+{
+  gchar *start;
+
+  g_return_if_fail (hdr != NULL);
+  g_return_if_fail (methods != NULL);
+  g_return_if_fail (stale != NULL);
+
+  /* Skip whitespace at the start of the string */
+  for (start = hdr; start[0] != '\0' && g_ascii_isspace (start[0]); start++);
+
+  if (g_ascii_strncasecmp (start, "basic", 5) == 0)
+    *methods |= GST_RTSP_AUTH_BASIC;
+  else if (g_ascii_strncasecmp (start, "digest ", 7) == 0) {
+    *methods |= GST_RTSP_AUTH_DIGEST;
+    gst_rtspsrc_parse_digest_challenge (conn, &start[7], stale);
+  }
+}
+
+/**
+ * gst_rtspsrc_setup_auth:
+ * @src: the rtsp source
+ *
+ * Configure a username and password and auth method on the
+ * connection object based on a response we received from the
+ * peer.
+ *
+ * Currently, this requires that a username and password were supplied
+ * in the uri. In the future, they may be requested on demand by sending
+ * a message up the bus.
+ *
+ * Returns: TRUE if authentication information could be set up correctly.
+ */
+static gboolean
+gst_rtspsrc_setup_auth (GstRTSPSrc * src, GstRTSPMessage * response)
+{
+  gchar *user = NULL;
+  gchar *pass = NULL;
+  GstRTSPAuthMethod avail_methods = GST_RTSP_AUTH_NONE;
+  GstRTSPAuthMethod method;
+  GstRTSPResult auth_result;
+  GstRTSPUrl *url;
+  GstRTSPConnection *conn;
+  gchar *hdr;
+  gboolean stale = FALSE;
+
+  conn = src->conninfo.connection;
+
+  /* Identify the available auth methods and see if any are supported */
+  if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_WWW_AUTHENTICATE,
+          &hdr, 0) == GST_RTSP_OK) {
+    gst_rtspsrc_parse_auth_hdr (hdr, &avail_methods, conn, &stale);
+  }
+
+  if (avail_methods == GST_RTSP_AUTH_NONE)
+    goto no_auth_available;
+
+  /* For digest auth, if the response indicates that the session
+   * data are stale, we just update them in the connection object and
+   * return TRUE to retry the request */
+  if (stale)
+    src->tried_url_auth = FALSE;
+
+  url = gst_rtsp_connection_get_url (conn);
+
+  /* Do we have username and password available? */
+  if (url != NULL && !src->tried_url_auth && url->user != NULL
+      && url->passwd != NULL) {
+    user = url->user;
+    pass = url->passwd;
+    src->tried_url_auth = TRUE;
+    GST_DEBUG_OBJECT (src,
+        "Attempting authentication using credentials from the URL");
+  } else {
+    user = src->user_id;
+    pass = src->user_pw;
+    GST_DEBUG_OBJECT (src,
+        "Attempting authentication using credentials from the properties");
+  }
+
+  /* FIXME: If the url didn't contain username and password or we tried them
+   * already, request a username and passwd from the application via some kind
+   * of credentials request message */
+
+  /* If we don't have a username and passwd at this point, bail out. */
+  if (user == NULL || pass == NULL)
+    goto no_user_pass;
+
+  /* Try to configure for each available authentication method, strongest to
+   * weakest */
+  for (method = GST_RTSP_AUTH_MAX; method != GST_RTSP_AUTH_NONE; method >>= 1) {
+    /* Check if this method is available on the server */
+    if ((method & avail_methods) == 0)
+      continue;
+
+    /* Pass the credentials to the connection to try on the next request */
+    auth_result = gst_rtsp_connection_set_auth (conn, method, user, pass);
+    /* INVAL indicates an invalid username/passwd were supplied, so we'll just
+     * ignore it and end up retrying later */
+    if (auth_result == GST_RTSP_OK || auth_result == GST_RTSP_EINVAL) {
+      GST_DEBUG_OBJECT (src, "Attempting %s authentication",
+          gst_rtsp_auth_method_to_string (method));
+      break;
+    }
+  }
+
+  if (method == GST_RTSP_AUTH_NONE)
+    goto no_auth_available;
+
+  return TRUE;
+
+no_auth_available:
+  {
+    /* Output an error indicating that we couldn't connect because there were
+     * no supported authentication protocols */
+    GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
+        ("No supported authentication protocol was found"));
+    return FALSE;
+  }
+no_user_pass:
+  {
+    /* We don't fire an error message, we just return FALSE and let the
+     * normal NOT_AUTHORIZED error be propagated */
+    return FALSE;
+  }
+}
+
+static GstRTSPResult
+gst_rtspsrc_try_send (GstRTSPSrc * src, GstRTSPConnection * conn,
+    GstRTSPMessage * request, GstRTSPMessage * response,
+    GstRTSPStatusCode * code)
+{
+  GstRTSPResult res;
+  GstRTSPStatusCode thecode;
+  gchar *content_base = NULL;
+  gint try = 0;
+
+again:
+  if (!src->short_header)
+    gst_rtsp_ext_list_before_send (src->extensions, request);
+
+  GST_DEBUG_OBJECT (src, "sending message");
+
+  if (src->debug)
+    gst_rtsp_message_dump (request);
+
+  res = gst_rtspsrc_connection_send (src, conn, request, src->ptcp_timeout);
+  if (res < 0)
+    goto send_error;
+
+  gst_rtsp_connection_reset_timeout (conn);
+
+next:
+  res = gst_rtspsrc_connection_receive (src, conn, response, src->ptcp_timeout);
+  if (res < 0)
+    goto receive_error;
+
+  if (src->debug)
+    gst_rtsp_message_dump (response);
+
+  switch (response->type) {
+    case GST_RTSP_MESSAGE_REQUEST:
+      res = gst_rtspsrc_handle_request (src, conn, response);
+      if (res == GST_RTSP_EEOF)
+        goto server_eof;
+      else if (res < 0)
+        goto handle_request_failed;
+      goto next;
+    case GST_RTSP_MESSAGE_RESPONSE:
+      /* ok, a response is good */
+      GST_DEBUG_OBJECT (src, "received response message");
+      break;
+    case GST_RTSP_MESSAGE_DATA:
+      /* get next response */
+      GST_DEBUG_OBJECT (src, "handle data response message");
+      gst_rtspsrc_handle_data (src, response);
+      goto next;
+    default:
+      GST_WARNING_OBJECT (src, "ignoring unknown message type %d",
+          response->type);
+      goto next;
+  }
+
+  thecode = response->type_data.response.code;
+
+  GST_DEBUG_OBJECT (src, "got response message %d", thecode);
+
+  /* if the caller wanted the result code, we store it. */
+  if (code)
+    *code = thecode;
+
+  /* If the request didn't succeed, bail out before doing any more */
+  if (thecode != GST_RTSP_STS_OK)
+    return GST_RTSP_OK;
+
+  /* store new content base if any */
+  gst_rtsp_message_get_header (response, GST_RTSP_HDR_CONTENT_BASE,
+      &content_base, 0);
+  if (content_base) {
+    g_free (src->content_base);
+    src->content_base = g_strdup (content_base);
+  }
+  gst_rtsp_ext_list_after_send (src->extensions, request, response);
+
+  return GST_RTSP_OK;
+
+  /* ERRORS */
+send_error:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    if (res != GST_RTSP_EINTR) {
+      GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
+          ("Could not send message. (%s)", str));
+    } else {
+      GST_WARNING_OBJECT (src, "send interrupted");
+    }
+    g_free (str);
+    return res;
+  }
+receive_error:
+  {
+    switch (res) {
+      case GST_RTSP_EEOF:
+        GST_WARNING_OBJECT (src, "server closed connection");
+        if ((try == 0) && !src->interleaved && src->udp_reconnect) {
+          try++;
+          /* if reconnect succeeds, try again */
+          if ((res =
+                  gst_rtsp_conninfo_reconnect (src, &src->conninfo,
+                      FALSE)) == 0)
+            goto again;
+        }
+        /* only try once after reconnect, then fallthrough and error out */
+      default:
+      {
+        gchar *str = gst_rtsp_strresult (res);
+
+        if (res != GST_RTSP_EINTR) {
+          GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
+              ("Could not receive message. (%s)", str));
+        } else {
+          GST_WARNING_OBJECT (src, "receive interrupted");
+        }
+        g_free (str);
+        break;
+      }
+    }
+    return res;
+  }
+handle_request_failed:
+  {
+    /* ERROR was posted */
+    gst_rtsp_message_unset (response);
+    return res;
+  }
+server_eof:
+  {
+    GST_DEBUG_OBJECT (src, "we got an eof from the server");
+    GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
+        ("The server closed the connection."));
+    gst_rtsp_message_unset (response);
+    return res;
+  }
+}
+
+/**
+ * gst_rtspsrc_send:
+ * @src: the rtsp source
+ * @conn: the connection to send on
+ * @request: must point to a valid request
+ * @response: must point to an empty #GstRTSPMessage
+ * @code: an optional code result
+ *
+ * send @request and retrieve the response in @response. optionally @code can be
+ * non-NULL in which case it will contain the status code of the response.
+ *
+ * If This function returns #GST_RTSP_OK, @response will contain a valid response
+ * message that should be cleaned with gst_rtsp_message_unset() after usage.
+ *
+ * If @code is NULL, this function will return #GST_RTSP_ERROR (with an invalid
+ * @response message) if the response code was not 200 (OK).
+ *
+ * If the attempt results in an authentication failure, then this will attempt
+ * to retrieve authentication credentials via gst_rtspsrc_setup_auth and retry
+ * the request.
+ *
+ * Returns: #GST_RTSP_OK if the processing was successful.
+ */
+static GstRTSPResult
+gst_rtspsrc_send (GstRTSPSrc * src, GstRTSPConnection * conn,
+    GstRTSPMessage * request, GstRTSPMessage * response,
+    GstRTSPStatusCode * code)
+{
+  GstRTSPStatusCode int_code = GST_RTSP_STS_OK;
+  GstRTSPResult res = GST_RTSP_ERROR;
+  gint count;
+  gboolean retry;
+  GstRTSPMethod method = GST_RTSP_INVALID;
+
+  count = 0;
+  do {
+    retry = FALSE;
+
+    /* make sure we don't loop forever */
+    if (count++ > 8)
+      break;
+
+    /* save method so we can disable it when the server complains */
+    method = request->type_data.request.method;
+
+    if ((res =
+            gst_rtspsrc_try_send (src, conn, request, response, &int_code)) < 0)
+      goto error;
+
+    switch (int_code) {
+      case GST_RTSP_STS_UNAUTHORIZED:
+        if (gst_rtspsrc_setup_auth (src, response)) {
+          /* Try the request/response again after configuring the auth info
+           * and loop again */
+          retry = TRUE;
+        }
+        break;
+      default:
+        break;
+    }
+  } while (retry == TRUE);
+
+  /* If the user requested the code, let them handle errors, otherwise
+   * post an error below */
+  if (code != NULL)
+    *code = int_code;
+  else if (int_code != GST_RTSP_STS_OK)
+    goto error_response;
+
+  return res;
+
+  /* ERRORS */
+error:
+  {
+    GST_DEBUG_OBJECT (src, "got error %d", res);
+    return res;
+  }
+error_response:
+  {
+    res = GST_RTSP_ERROR;
+
+    switch (response->type_data.response.code) {
+      case GST_RTSP_STS_NOT_FOUND:
+        GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL), ("%s",
+                response->type_data.response.reason));
+        break;
+      case GST_RTSP_STS_UNAUTHORIZED:
+        GST_ELEMENT_ERROR (src, RESOURCE, NOT_AUTHORIZED, (NULL), ("%s",
+                response->type_data.response.reason));
+        break;
+      case GST_RTSP_STS_MOVED_PERMANENTLY:
+      case GST_RTSP_STS_MOVE_TEMPORARILY:
+      {
+        gchar *new_location;
+        GstRTSPLowerTrans transports;
+
+        GST_DEBUG_OBJECT (src, "got redirection");
+        /* if we don't have a Location Header, we must error */
+        if (gst_rtsp_message_get_header (response, GST_RTSP_HDR_LOCATION,
+                &new_location, 0) < 0)
+          break;
+
+        /* When we receive a redirect result, we go back to the INIT state after
+         * parsing the new URI. The caller should do the needed steps to issue
+         * a new setup when it detects this state change. */
+        GST_DEBUG_OBJECT (src, "redirection to %s", new_location);
+
+        /* save current transports */
+        if (src->conninfo.url)
+          transports = src->conninfo.url->transports;
+        else
+          transports = GST_RTSP_LOWER_TRANS_UNKNOWN;
+
+        gst_rtspsrc_uri_set_uri (GST_URI_HANDLER (src), new_location, NULL);
+
+        /* set old transports */
+        if (src->conninfo.url && transports != GST_RTSP_LOWER_TRANS_UNKNOWN)
+          src->conninfo.url->transports = transports;
+
+        src->need_redirect = TRUE;
+        src->state = GST_RTSP_STATE_INIT;
+        res = GST_RTSP_OK;
+        break;
+      }
+      case GST_RTSP_STS_NOT_ACCEPTABLE:
+      case GST_RTSP_STS_NOT_IMPLEMENTED:
+      case GST_RTSP_STS_METHOD_NOT_ALLOWED:
+        GST_WARNING_OBJECT (src, "got NOT IMPLEMENTED, disable method %s",
+            gst_rtsp_method_as_text (method));
+        src->methods &= ~method;
+        res = GST_RTSP_OK;
+        break;
+      default:
+        GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
+            ("Got error response: %d (%s).", response->type_data.response.code,
+                response->type_data.response.reason));
+        break;
+    }
+    /* if we return ERROR we should unset the response ourselves */
+    if (res == GST_RTSP_ERROR)
+      gst_rtsp_message_unset (response);
+
+    return res;
+  }
+}
+
+static GstRTSPResult
+gst_rtspsrc_send_cb (GstRTSPExtension * ext, GstRTSPMessage * request,
+    GstRTSPMessage * response, GstRTSPSrc * src)
+{
+  return gst_rtspsrc_send (src, src->conninfo.connection, request, response,
+      NULL);
+}
+
+
+/* parse the response and collect all the supported methods. We need this
+ * information so that we don't try to send an unsupported request to the
+ * server.
+ */
+static gboolean
+gst_rtspsrc_parse_methods (GstRTSPSrc * src, GstRTSPMessage * response)
+{
+  GstRTSPHeaderField field;
+  gchar *respoptions;
+  gint indx = 0;
+
+  /* reset supported methods */
+  src->methods = 0;
+
+  /* Try Allow Header first */
+  field = GST_RTSP_HDR_ALLOW;
+  while (TRUE) {
+    respoptions = NULL;
+    gst_rtsp_message_get_header (response, field, &respoptions, indx);
+    if (indx == 0 && !respoptions) {
+      /* if no Allow header was found then try the Public header... */
+      field = GST_RTSP_HDR_PUBLIC;
+      gst_rtsp_message_get_header (response, field, &respoptions, indx);
+    }
+    if (!respoptions)
+      break;
+
+    src->methods |= gst_rtsp_options_from_text (respoptions);
+
+    indx++;
+  }
+
+  if (src->methods == 0) {
+    /* neither Allow nor Public are required, assume the server supports
+     * at least DESCRIBE, SETUP, we always assume it supports PLAY as
+     * well. */
+    GST_DEBUG_OBJECT (src, "could not get OPTIONS");
+    src->methods = GST_RTSP_DESCRIBE | GST_RTSP_SETUP;
+  }
+  /* always assume PLAY, FIXME, extensions should be able to override
+   * this */
+  src->methods |= GST_RTSP_PLAY;
+  /* also assume it will support Range */
+  src->seekable = TRUE;
+
+  /* we need describe and setup */
+  if (!(src->methods & GST_RTSP_DESCRIBE))
+    goto no_describe;
+  if (!(src->methods & GST_RTSP_SETUP))
+    goto no_setup;
+
+  return TRUE;
+
+  /* ERRORS */
+no_describe:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
+        ("Server does not support DESCRIBE."));
+    return FALSE;
+  }
+no_setup:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
+        ("Server does not support SETUP."));
+    return FALSE;
+  }
+}
+
+/* masks to be kept in sync with the hardcoded protocol order of preference
+ * in code below */
+static const guint protocol_masks[] = {
+  GST_RTSP_LOWER_TRANS_UDP,
+  GST_RTSP_LOWER_TRANS_UDP_MCAST,
+  GST_RTSP_LOWER_TRANS_TCP,
+  0
+};
+
+static GstRTSPResult
+gst_rtspsrc_create_transports_string (GstRTSPSrc * src,
+    GstRTSPLowerTrans protocols, GstRTSPProfile profile, gchar ** transports)
+{
+  GstRTSPResult res;
+  GString *result;
+  gboolean add_udp_str;
+
+  *transports = NULL;
+
+  res =
+      gst_rtsp_ext_list_get_transports (src->extensions, protocols, transports);
+
+  if (res < 0)
+    goto failed;
+
+  GST_DEBUG_OBJECT (src, "got transports %s", GST_STR_NULL (*transports));
+
+  /* extension listed transports, use those */
+  if (*transports != NULL)
+    return GST_RTSP_OK;
+
+  /* it's the default */
+  add_udp_str = FALSE;
+
+  /* the default RTSP transports */
+  result = g_string_new ("RTP");
+
+  switch (profile) {
+    case GST_RTSP_PROFILE_AVP:
+      g_string_append (result, "/AVP");
+      break;
+    case GST_RTSP_PROFILE_SAVP:
+      g_string_append (result, "/SAVP");
+      break;
+    case GST_RTSP_PROFILE_AVPF:
+      g_string_append (result, "/AVPF");
+      break;
+    case GST_RTSP_PROFILE_SAVPF:
+      g_string_append (result, "/SAVPF");
+      break;
+    default:
+      break;
+  }
+
+  if (protocols & GST_RTSP_LOWER_TRANS_UDP) {
+    GST_DEBUG_OBJECT (src, "adding UDP unicast");
+    if (add_udp_str)
+      g_string_append (result, "/UDP");
+    g_string_append (result, ";unicast;client_port=%%u1-%%u2");
+  } else if (protocols & GST_RTSP_LOWER_TRANS_UDP_MCAST) {
+    GST_DEBUG_OBJECT (src, "adding UDP multicast");
+    /* we don't have to allocate any UDP ports yet, if the selected transport
+     * turns out to be multicast we can create them and join the multicast
+     * group indicated in the transport reply */
+    if (add_udp_str)
+      g_string_append (result, "/UDP");
+    g_string_append (result, ";multicast");
+    if (src->next_port_num != 0) {
+      if (src->client_port_range.max > 0 &&
+          src->next_port_num >= src->client_port_range.max)
+        goto no_ports;
+
+      g_string_append_printf (result, ";client_port=%d-%d",
+          src->next_port_num, src->next_port_num + 1);
+    }
+  } else if (protocols & GST_RTSP_LOWER_TRANS_TCP) {
+    GST_DEBUG_OBJECT (src, "adding TCP");
+
+    g_string_append (result, "/TCP;unicast;interleaved=%%i1-%%i2");
+  }
+  *transports = g_string_free (result, FALSE);
+
+  GST_DEBUG_OBJECT (src, "prepared transports %s", GST_STR_NULL (*transports));
+
+  return GST_RTSP_OK;
+
+  /* ERRORS */
+failed:
+  {
+    GST_ERROR ("extension gave error %d", res);
+    return res;
+  }
+no_ports:
+  {
+    GST_ERROR ("no more ports available");
+    return GST_RTSP_ERROR;
+  }
+}
+
+static GstRTSPResult
+gst_rtspsrc_prepare_transports (GstRTSPStream * stream, gchar ** transports,
+    gint orig_rtpport, gint orig_rtcpport)
+{
+  GstRTSPSrc *src;
+  gint nr_udp, nr_int;
+  gchar *next, *p;
+  gint rtpport = 0, rtcpport = 0;
+  GString *str;
+
+  src = stream->parent;
+
+  /* find number of placeholders first */
+  if (strstr (*transports, "%%i2"))
+    nr_int = 2;
+  else if (strstr (*transports, "%%i1"))
+    nr_int = 1;
+  else
+    nr_int = 0;
+
+  if (strstr (*transports, "%%u2"))
+    nr_udp = 2;
+  else if (strstr (*transports, "%%u1"))
+    nr_udp = 1;
+  else
+    nr_udp = 0;
+
+  if (nr_udp == 0 && nr_int == 0)
+    goto done;
+
+  if (nr_udp > 0) {
+    if (!orig_rtpport || !orig_rtcpport) {
+      if (!gst_rtspsrc_alloc_udp_ports (stream, &rtpport, &rtcpport))
+        goto failed;
+    } else {
+      rtpport = orig_rtpport;
+      rtcpport = orig_rtcpport;
+    }
+  }
+
+  str = g_string_new ("");
+  p = *transports;
+  while ((next = strstr (p, "%%"))) {
+    g_string_append_len (str, p, next - p);
+    if (next[2] == 'u') {
+      if (next[3] == '1')
+        g_string_append_printf (str, "%d", rtpport);
+      else if (next[3] == '2')
+        g_string_append_printf (str, "%d", rtcpport);
+    }
+    if (next[2] == 'i') {
+      if (next[3] == '1')
+        g_string_append_printf (str, "%d", src->free_channel);
+      else if (next[3] == '2')
+        g_string_append_printf (str, "%d", src->free_channel + 1);
+    }
+
+    p = next + 4;
+  }
+  /* append final part */
+  g_string_append (str, p);
+
+  g_free (*transports);
+  *transports = g_string_free (str, FALSE);
+
+done:
+  return GST_RTSP_OK;
+
+  /* ERRORS */
+failed:
+  {
+    GST_ERROR ("failed to allocate udp ports");
+    return GST_RTSP_ERROR;
+  }
+}
+
+static guint8
+enc_key_length_from_cipher_name (const gchar * cipher)
+{
+  if (g_strcmp0 (cipher, "aes-128-icm") == 0)
+    return AES_128_KEY_LEN;
+  else if (g_strcmp0 (cipher, "aes-256-icm") == 0)
+    return AES_256_KEY_LEN;
+  else {
+    GST_ERROR ("encryption algorithm '%s' not supported", cipher);
+    return 0;
+  }
+}
+
+static guint8
+auth_key_length_from_auth_name (const gchar * auth)
+{
+  if (g_strcmp0 (auth, "hmac-sha1-32") == 0)
+    return HMAC_32_KEY_LEN;
+  else if (g_strcmp0 (auth, "hmac-sha1-80") == 0)
+    return HMAC_80_KEY_LEN;
+  else {
+    GST_ERROR ("authentication algorithm '%s' not supported", auth);
+    return 0;
+  }
+}
+
+static GstCaps *
+signal_get_srtcp_params (GstRTSPSrc * src, GstRTSPStream * stream)
+{
+  GstCaps *caps = NULL;
+
+  g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_REQUEST_RTCP_KEY], 0,
+      stream->id, &caps);
+
+  if (caps != NULL)
+    GST_DEBUG_OBJECT (src, "SRTP parameters received");
+
+  return caps;
+}
+
+static GstCaps *
+default_srtcp_params (void)
+{
+  guint i;
+  GstCaps *caps;
+  GstBuffer *buf;
+  guint8 *key_data;
+#define KEY_SIZE 30
+
+  /* create a random key */
+  key_data = g_malloc (KEY_SIZE);
+  for (i = 0; i < KEY_SIZE; i += 4)
+    GST_WRITE_UINT32_BE (key_data + i, g_random_int ());
+
+  buf = gst_buffer_new_wrapped (key_data, KEY_SIZE);
+
+  caps = gst_caps_new_simple ("application/x-srtp",
+      "srtp-key", GST_TYPE_BUFFER, buf,
+      "srtcp-cipher", G_TYPE_STRING, "aes-128-icm",
+      "srtcp-auth", G_TYPE_STRING, "hmac-sha1-80", NULL);
+
+  gst_buffer_unref (buf);
+
+  return caps;
+}
+
+static gchar *
+gst_rtspsrc_stream_make_keymgmt (GstRTSPSrc * src, GstRTSPStream * stream)
+{
+  GBytes *bytes;
+  gchar *result, *base64;
+  const guint8 *data;
+  gsize size;
+  GstMIKEYMessage *msg;
+  GstMIKEYPayload *payload, *pkd;
+  guint8 byte;
+  GstStructure *s;
+  GstMapInfo info;
+  GstBuffer *srtpkey;
+  const GValue *val;
+  const gchar *srtcpcipher, *srtcpauth;
+
+  stream->srtcpparams = signal_get_srtcp_params (src, stream);
+  if (stream->srtcpparams == NULL)
+    stream->srtcpparams = default_srtcp_params ();
+
+  s = gst_caps_get_structure (stream->srtcpparams, 0);
+
+  srtcpcipher = gst_structure_get_string (s, "srtcp-cipher");
+  srtcpauth = gst_structure_get_string (s, "srtcp-auth");
+  val = gst_structure_get_value (s, "srtp-key");
+
+  if (srtcpcipher == NULL || srtcpauth == NULL || val == NULL) {
+    GST_ERROR_OBJECT (src, "could not find the right SRTP parameters in caps");
+    return NULL;
+  }
+
+  srtpkey = gst_value_get_buffer (val);
+
+  msg = gst_mikey_message_new ();
+  /* unencrypted MIKEY message, we send this over TLS so this is allowed */
+  gst_mikey_message_set_info (msg, GST_MIKEY_VERSION, GST_MIKEY_TYPE_PSK_INIT,
+      FALSE, GST_MIKEY_PRF_MIKEY_1, g_random_int (), GST_MIKEY_MAP_TYPE_SRTP);
+  /* add policy '0' for our SSRC */
+  gst_mikey_message_add_cs_srtp (msg, 0, stream->send_ssrc, 0);
+  /* timestamp is now */
+  gst_mikey_message_add_t_now_ntp_utc (msg);
+  /* add some random data */
+  gst_mikey_message_add_rand_len (msg, 16);
+
+  /* the policy '0' is SRTP */
+  payload = gst_mikey_payload_new (GST_MIKEY_PT_SP);
+  gst_mikey_payload_sp_set (payload, 0, GST_MIKEY_SEC_PROTO_SRTP);
+
+  /* only AES-CM is supported */
+  byte = 1;
+  gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_ALG, 1, &byte);
+  /* encryption key length */
+  byte = enc_key_length_from_cipher_name (srtcpcipher);
+  gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_ENC_KEY_LEN, 1,
+      &byte);
+  /* only HMAC-SHA1 */
+  gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_ALG, 1,
+      &byte);
+  /* authentication key length */
+  byte = auth_key_length_from_auth_name (srtcpauth);
+  gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_AUTH_KEY_LEN, 1,
+      &byte);
+  /* we enable encryption on RTP and RTCP */
+  gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_ENC, 1,
+      &byte);
+  gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTCP_ENC, 1,
+      &byte);
+  /* we enable authentication on RTP and RTCP */
+  gst_mikey_payload_sp_add_param (payload, GST_MIKEY_SP_SRTP_SRTP_AUTH, 1,
+      &byte);
+  gst_mikey_message_add_payload (msg, payload);
+
+  /* make unencrypted KEMAC */
+  payload = gst_mikey_payload_new (GST_MIKEY_PT_KEMAC);
+  gst_mikey_payload_kemac_set (payload, GST_MIKEY_ENC_NULL, GST_MIKEY_MAC_NULL);
+  /* add the key in KEMAC */
+  pkd = gst_mikey_payload_new (GST_MIKEY_PT_KEY_DATA);
+  gst_buffer_map (srtpkey, &info, GST_MAP_READ);
+  gst_mikey_payload_key_data_set_key (pkd, GST_MIKEY_KD_TEK, info.size,
+      info.data);
+  gst_buffer_unmap (srtpkey, &info);
+  gst_mikey_payload_kemac_add_sub (payload, pkd);
+  gst_mikey_message_add_payload (msg, payload);
+
+  /* now serialize this to bytes */
+  bytes = gst_mikey_message_to_bytes (msg, NULL, NULL);
+  gst_mikey_message_unref (msg);
+  /* and make it into base64 */
+  data = g_bytes_get_data (bytes, &size);
+  base64 = g_base64_encode (data, size);
+  g_bytes_unref (bytes);
+
+  result = g_strdup_printf ("prot=mikey;uri=\"%s\";data=\"%s\"",
+      stream->conninfo.location, base64);
+  g_free (base64);
+
+  return result;
+}
+
+
+/* Perform the SETUP request for all the streams.
+ *
+ * We ask the server for a specific transport, which initially includes all the
+ * ones we can support (UDP/TCP/MULTICAST). For the UDP transport we allocate
+ * two local UDP ports that we send to the server.
+ *
+ * Once the server replied with a transport, we configure the other streams
+ * with the same transport.
+ *
+ * This function will also configure the stream for the selected transport,
+ * which basically means creating the pipeline.
+ */
+static GstRTSPResult
+gst_rtspsrc_setup_streams (GstRTSPSrc * src, gboolean async)
+{
+  GList *walk;
+  GstRTSPResult res = GST_RTSP_ERROR;
+  GstRTSPMessage request = { 0 };
+  GstRTSPMessage response = { 0 };
+  GstRTSPStream *stream = NULL;
+  GstRTSPLowerTrans protocols;
+  GstRTSPStatusCode code;
+  gboolean unsupported_real = FALSE;
+  gint rtpport, rtcpport;
+  GstRTSPUrl *url;
+  gchar *hval;
+
+  if (src->conninfo.connection) {
+    url = gst_rtsp_connection_get_url (src->conninfo.connection);
+    /* we initially allow all configured lower transports. based on the URL
+     * transports and the replies from the server we narrow them down. */
+    protocols = url->transports & src->cur_protocols;
+  } else {
+    url = NULL;
+    protocols = src->cur_protocols;
+  }
+
+  if (protocols == 0)
+    goto no_protocols;
+
+  /* reset some state */
+  src->free_channel = 0;
+  src->interleaved = FALSE;
+  src->need_activate = FALSE;
+  /* keep track of next port number, 0 is random */
+  src->next_port_num = src->client_port_range.min;
+  rtpport = rtcpport = 0;
+
+  if (G_UNLIKELY (src->streams == NULL))
+    goto no_streams;
+
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPConnection *conn;
+    gchar *transports;
+    gint retry = 0;
+    guint mask = 0;
+    gboolean selected;
+    GstCaps *caps;
+
+    stream = (GstRTSPStream *) walk->data;
+
+    caps = stream_get_caps_for_pt (stream, stream->default_pt);
+    if (caps == NULL) {
+      GST_DEBUG_OBJECT (src, "skipping stream %p, no caps", stream);
+      continue;
+    }
+
+    if (stream->skipped) {
+      GST_DEBUG_OBJECT (src, "skipping stream %p", stream);
+      continue;
+    }
+
+    /* see if we need to configure this stream */
+    if (!gst_rtsp_ext_list_configure_stream (src->extensions, caps)) {
+      GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by extension",
+          stream);
+      continue;
+    }
+
+    g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_SELECT_STREAM], 0,
+        stream->id, caps, &selected);
+    if (!selected) {
+      GST_DEBUG_OBJECT (src, "skipping stream %p, disabled by signal", stream);
+      continue;
+    }
+
+    /* merge/overwrite global caps */
+    if (caps) {
+      guint j, num;
+      GstStructure *s;
+
+      s = gst_caps_get_structure (caps, 0);
+
+      num = gst_structure_n_fields (src->props);
+      for (j = 0; j < num; j++) {
+        const gchar *name;
+        const GValue *val;
+
+        name = gst_structure_nth_field_name (src->props, j);
+        val = gst_structure_get_value (src->props, name);
+        gst_structure_set_value (s, name, val);
+
+        GST_DEBUG_OBJECT (src, "copied %s", name);
+      }
+    }
+
+    /* skip setup if we have no URL for it */
+    if (stream->conninfo.location == NULL) {
+      GST_DEBUG_OBJECT (src, "skipping stream %p, no setup", stream);
+      continue;
+    }
+
+    if (src->conninfo.connection == NULL) {
+      if (!gst_rtsp_conninfo_connect (src, &stream->conninfo, async)) {
+        GST_DEBUG_OBJECT (src, "skipping stream %p, failed to connect", stream);
+        continue;
+      }
+      conn = stream->conninfo.connection;
+    } else {
+      conn = src->conninfo.connection;
+    }
+    GST_DEBUG_OBJECT (src, "doing setup of stream %p with %s", stream,
+        stream->conninfo.location);
+
+    /* if we have a multicast connection, only suggest multicast from now on */
+    if (stream->is_multicast)
+      protocols &= GST_RTSP_LOWER_TRANS_UDP_MCAST;
+
+  next_protocol:
+    /* first selectable protocol */
+    while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
+      mask++;
+    if (!protocol_masks[mask])
+      goto no_protocols;
+
+  retry:
+    GST_DEBUG_OBJECT (src, "protocols = 0x%x, protocol mask = 0x%x", protocols,
+        protocol_masks[mask]);
+    /* create a string with first transport in line */
+    transports = NULL;
+    res = gst_rtspsrc_create_transports_string (src,
+        protocols & protocol_masks[mask], stream->profile, &transports);
+    if (res < 0 || transports == NULL)
+      goto setup_transport_failed;
+
+    if (strlen (transports) == 0) {
+      g_free (transports);
+      GST_DEBUG_OBJECT (src, "no transports found");
+      mask++;
+      goto next_protocol;
+    }
+
+    GST_DEBUG_OBJECT (src, "replace ports in %s", GST_STR_NULL (transports));
+
+    /* replace placeholders with real values, this function will optionally
+     * allocate UDP ports and other info needed to execute the setup request */
+    res = gst_rtspsrc_prepare_transports (stream, &transports,
+        retry > 0 ? rtpport : 0, retry > 0 ? rtcpport : 0);
+    if (res < 0) {
+      g_free (transports);
+      goto setup_transport_failed;
+    }
+
+    GST_DEBUG_OBJECT (src, "transport is now %s", GST_STR_NULL (transports));
+
+    /* create SETUP request */
+    res =
+        gst_rtsp_message_init_request (&request, GST_RTSP_SETUP,
+        stream->conninfo.location);
+    if (res < 0) {
+      g_free (transports);
+      goto create_request_failed;
+    }
+
+    /* select transport */
+    gst_rtsp_message_take_header (&request, GST_RTSP_HDR_TRANSPORT, transports);
+
+    /* set up keys */
+    if (stream->profile == GST_RTSP_PROFILE_SAVP ||
+        stream->profile == GST_RTSP_PROFILE_SAVPF) {
+      hval = gst_rtspsrc_stream_make_keymgmt (src, stream);
+      gst_rtsp_message_take_header (&request, GST_RTSP_HDR_KEYMGMT, hval);
+    }
+
+    /* if the user wants a non default RTP packet size we add the blocksize
+     * parameter */
+    if (src->rtp_blocksize > 0) {
+      hval = g_strdup_printf ("%d", src->rtp_blocksize);
+      gst_rtsp_message_take_header (&request, GST_RTSP_HDR_BLOCKSIZE, hval);
+    }
+
+    if (async)
+      GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("SETUP stream %d",
+              stream->id));
+
+    /* handle the code ourselves */
+    res = gst_rtspsrc_send (src, conn, &request, &response, &code);
+    if (res < 0)
+      goto send_error;
+
+    switch (code) {
+      case GST_RTSP_STS_OK:
+        break;
+      case GST_RTSP_STS_UNSUPPORTED_TRANSPORT:
+        gst_rtsp_message_unset (&request);
+        gst_rtsp_message_unset (&response);
+        /* cleanup of leftover transport */
+        gst_rtspsrc_stream_free_udp (stream);
+        /* MS WMServer RTSP MUST use same UDP pair in all SETUP requests;
+         * we might be in this case */
+        if (stream->container && rtpport && rtcpport && !retry) {
+          GST_DEBUG_OBJECT (src, "retrying with original port pair %u-%u",
+              rtpport, rtcpport);
+          retry++;
+          goto retry;
+        }
+        /* this transport did not go down well, but we may have others to try
+         * that we did not send yet, try those and only give up then
+         * but not without checking for lost cause/extension so we can
+         * post a nicer/more useful error message later */
+        if (!unsupported_real)
+          unsupported_real = stream->is_real;
+        /* select next available protocol, give up on this stream if none */
+        mask++;
+        while (protocol_masks[mask] && !(protocols & protocol_masks[mask]))
+          mask++;
+        if (!protocol_masks[mask] || unsupported_real)
+          continue;
+        else
+          goto retry;
+      default:
+        /* cleanup of leftover transport and move to the next stream */
+        gst_rtspsrc_stream_free_udp (stream);
+        goto response_error;
+    }
+
+    /* parse response transport */
+    {
+      gchar *resptrans = NULL;
+      GstRTSPTransport transport = { 0 };
+
+      gst_rtsp_message_get_header (&response, GST_RTSP_HDR_TRANSPORT,
+          &resptrans, 0);
+      if (!resptrans) {
+        gst_rtspsrc_stream_free_udp (stream);
+        goto no_transport;
+      }
+
+      /* parse transport, go to next stream on parse error */
+      if (gst_rtsp_transport_parse (resptrans, &transport) != GST_RTSP_OK) {
+        GST_WARNING_OBJECT (src, "failed to parse transport %s", resptrans);
+        goto next;
+      }
+
+      /* update allowed transports for other streams. once the transport of
+       * one stream has been determined, we make sure that all other streams
+       * are configured in the same way */
+      switch (transport.lower_transport) {
+        case GST_RTSP_LOWER_TRANS_TCP:
+          GST_DEBUG_OBJECT (src, "stream %p as TCP interleaved", stream);
+          protocols = GST_RTSP_LOWER_TRANS_TCP;
+          src->interleaved = TRUE;
+          /* update free channels */
+          src->free_channel =
+              MAX (transport.interleaved.min, src->free_channel);
+          src->free_channel =
+              MAX (transport.interleaved.max, src->free_channel);
+          src->free_channel++;
+          break;
+        case GST_RTSP_LOWER_TRANS_UDP_MCAST:
+          /* only allow multicast for other streams */
+          GST_DEBUG_OBJECT (src, "stream %p as UDP multicast", stream);
+          protocols = GST_RTSP_LOWER_TRANS_UDP_MCAST;
+          /* if the server selected our ports, increment our counters so that
+           * we select a new port later */
+          if (src->next_port_num == transport.port.min &&
+              src->next_port_num + 1 == transport.port.max) {
+            src->next_port_num += 2;
+          }
+          break;
+        case GST_RTSP_LOWER_TRANS_UDP:
+          /* only allow unicast for other streams */
+          GST_DEBUG_OBJECT (src, "stream %p as UDP unicast", stream);
+          protocols = GST_RTSP_LOWER_TRANS_UDP;
+          break;
+        default:
+          GST_DEBUG_OBJECT (src, "stream %p unknown transport %d", stream,
+              transport.lower_transport);
+          break;
+      }
+
+      if (!src->interleaved || !retry) {
+        /* now configure the stream with the selected transport */
+        if (!gst_rtspsrc_stream_configure_transport (stream, &transport)) {
+          GST_DEBUG_OBJECT (src,
+              "could not configure stream %p transport, skipping stream",
+              stream);
+          goto next;
+        } else if (stream->udpsrc[0] && stream->udpsrc[1]) {
+          /* retain the first allocated UDP port pair */
+          g_object_get (G_OBJECT (stream->udpsrc[0]), "port", &rtpport, NULL);
+          g_object_get (G_OBJECT (stream->udpsrc[1]), "port", &rtcpport, NULL);
+        }
+      }
+      /* we need to activate at least one streams when we detect activity */
+      src->need_activate = TRUE;
+
+      /* stream is setup now */
+      stream->setup = TRUE;
+      {
+        GList *skip = walk;
+
+        while (TRUE) {
+          GstRTSPStream *sskip;
+
+          skip = g_list_next (skip);
+          if (skip == NULL)
+            break;
+
+          sskip = (GstRTSPStream *) skip->data;
+
+          /* skip all streams with the same control url */
+          if (g_str_equal (stream->conninfo.location, sskip->conninfo.location)) {
+            GST_DEBUG_OBJECT (src, "found stream %p with same control %s",
+                sskip, sskip->conninfo.location);
+            sskip->skipped = TRUE;
+          }
+        }
+      }
+    next:
+      /* clean up our transport struct */
+      gst_rtsp_transport_init (&transport);
+      /* clean up used RTSP messages */
+      gst_rtsp_message_unset (&request);
+      gst_rtsp_message_unset (&response);
+    }
+  }
+
+  /* store the transport protocol that was configured */
+  src->cur_protocols = protocols;
+
+  gst_rtsp_ext_list_stream_select (src->extensions, url);
+
+  /* if there is nothing to activate, error out */
+  if (!src->need_activate)
+    goto nothing_to_activate;
+
+  return res;
+
+  /* ERRORS */
+no_protocols:
+  {
+    /* no transport possible, post an error and stop */
+    GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
+        ("Could not connect to server, no protocols left"));
+    return GST_RTSP_ERROR;
+  }
+no_streams:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
+        ("SDP contains no streams"));
+    return GST_RTSP_ERROR;
+  }
+create_request_failed:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
+        ("Could not create request. (%s)", str));
+    g_free (str);
+    goto cleanup_error;
+  }
+setup_transport_failed:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
+        ("Could not setup transport."));
+    res = GST_RTSP_ERROR;
+    goto cleanup_error;
+  }
+response_error:
+  {
+    const gchar *str = gst_rtsp_status_as_text (code);
+
+    GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
+        ("Error (%d): %s", code, GST_STR_NULL (str)));
+    res = GST_RTSP_ERROR;
+    goto cleanup_error;
+  }
+send_error:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    if (res != GST_RTSP_EINTR) {
+      GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
+          ("Could not send message. (%s)", str));
+    } else {
+      GST_WARNING_OBJECT (src, "send interrupted");
+    }
+    g_free (str);
+    goto cleanup_error;
+  }
+no_transport:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
+        ("Server did not select transport."));
+    res = GST_RTSP_ERROR;
+    goto cleanup_error;
+  }
+nothing_to_activate:
+  {
+    /* none of the available error codes is really right .. */
+    if (unsupported_real) {
+      GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
+          (_("No supported stream was found. You might need to install a "
+                  "GStreamer RTSP extension plugin for Real media streams.")),
+          (NULL));
+    } else {
+      GST_ELEMENT_ERROR (src, STREAM, CODEC_NOT_FOUND,
+          (_("No supported stream was found. You might need to allow "
+                  "more transport protocols or may otherwise be missing "
+                  "the right GStreamer RTSP extension plugin.")), (NULL));
+    }
+    return GST_RTSP_ERROR;
+  }
+cleanup_error:
+  {
+    gst_rtsp_message_unset (&request);
+    gst_rtsp_message_unset (&response);
+    return res;
+  }
+}
+
+static gboolean
+gst_rtspsrc_parse_range (GstRTSPSrc * src, const gchar * range,
+    GstSegment * segment)
+{
+  gint64 seconds;
+  GstRTSPTimeRange *therange;
+
+  if (src->range)
+    gst_rtsp_range_free (src->range);
+
+  if (gst_rtsp_range_parse (range, &therange) == GST_RTSP_OK) {
+    GST_DEBUG_OBJECT (src, "parsed range %s", range);
+    src->range = therange;
+  } else {
+    GST_DEBUG_OBJECT (src, "failed to parse range %s", range);
+    src->range = NULL;
+    gst_segment_init (segment, GST_FORMAT_TIME);
+    return FALSE;
+  }
+
+  GST_DEBUG_OBJECT (src, "range: type %d, min %f - type %d,  max %f ",
+      therange->min.type, therange->min.seconds, therange->max.type,
+      therange->max.seconds);
+
+  if (therange->min.type == GST_RTSP_TIME_NOW)
+    seconds = 0;
+  else if (therange->min.type == GST_RTSP_TIME_END)
+    seconds = 0;
+  else
+    seconds = therange->min.seconds * GST_SECOND;
+
+  GST_DEBUG_OBJECT (src, "range: min %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (seconds));
+
+  /* we need to start playback without clipping from the position reported by
+   * the server */
+  segment->start = seconds;
+  segment->position = seconds;
+
+  if (therange->max.type == GST_RTSP_TIME_NOW)
+    seconds = -1;
+  else if (therange->max.type == GST_RTSP_TIME_END)
+    seconds = -1;
+  else
+    seconds = therange->max.seconds * GST_SECOND;
+
+  GST_DEBUG_OBJECT (src, "range: max %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (seconds));
+
+  /* live (WMS) server might send overflowed large max as its idea of infinity,
+   * compensate to prevent problems later on */
+  if (seconds != -1 && seconds < 0) {
+    seconds = -1;
+    GST_DEBUG_OBJECT (src, "insane range, set to NONE");
+  }
+
+  /* live (WMS) might send min == max, which is not worth recording */
+  if (segment->duration == -1 && seconds == segment->start)
+    seconds = -1;
+
+  /* don't change duration with unknown value, we might have a valid value
+   * there that we want to keep. */
+  if (seconds != -1)
+    segment->duration = seconds;
+
+  return TRUE;
+}
+
+/* Parse clock profived by the server with following syntax:
+ *
+ * "GstNetTimeProvider <wrapped-clock> <server-IP:port> <clock-time>"
+ */
+static gboolean
+gst_rtspsrc_parse_gst_clock (GstRTSPSrc * src, const gchar * gstclock)
+{
+  gboolean res = FALSE;
+
+  if (g_str_has_prefix (gstclock, "GstNetTimeProvider ")) {
+    gchar **fields = NULL, **parts = NULL;
+    gchar *remote_ip, *str;
+    gint port;
+    GstClockTime base_time;
+    GstClock *netclock;
+
+    fields = g_strsplit (gstclock, " ", 0);
+
+    /* wrapped clock, not very interesting for now */
+    if (fields[1] == NULL)
+      goto cleanup;
+
+    /* remote IP address and port */
+    if ((str = fields[2]) == NULL)
+      goto cleanup;
+
+    parts = g_strsplit (str, ":", 0);
+
+    if ((remote_ip = parts[0]) == NULL)
+      goto cleanup;
+
+    if ((str = parts[1]) == NULL)
+      goto cleanup;
+
+    port = atoi (str);
+    if (port == 0)
+      goto cleanup;
+
+    /* base-time */
+    if ((str = fields[3]) == NULL)
+      goto cleanup;
+
+    base_time = g_ascii_strtoull (str, NULL, 10);
+
+    netclock =
+        gst_net_client_clock_new ((gchar *) "GstRTSPClock", remote_ip, port,
+        base_time);
+
+    if (src->provided_clock)
+      gst_object_unref (src->provided_clock);
+    src->provided_clock = netclock;
+
+    gst_element_post_message (GST_ELEMENT_CAST (src),
+        gst_message_new_clock_provide (GST_OBJECT_CAST (src),
+            src->provided_clock, TRUE));
+
+    res = TRUE;
+  cleanup:
+    g_strfreev (fields);
+    g_strfreev (parts);
+  }
+  return res;
+}
+
+/* must be called with the RTSP state lock */
+static GstRTSPResult
+gst_rtspsrc_open_from_sdp (GstRTSPSrc * src, GstSDPMessage * sdp,
+    gboolean async)
+{
+  GstRTSPResult res;
+  gint i, n_streams;
+
+  /* prepare global stream caps properties */
+  if (src->props)
+    gst_structure_remove_all_fields (src->props);
+  else
+    src->props = gst_structure_new_empty ("RTSPProperties");
+
+  if (src->debug)
+    gst_sdp_message_dump (sdp);
+
+  gst_rtsp_ext_list_parse_sdp (src->extensions, sdp, src->props);
+
+  /* let the app inspect and change the SDP */
+  g_signal_emit (src, gst_rtspsrc_signals[SIGNAL_ON_SDP], 0, sdp);
+
+  gst_segment_init (&src->segment, GST_FORMAT_TIME);
+
+  /* parse range for duration reporting. */
+  {
+    const gchar *range;
+
+    for (i = 0;; i++) {
+      range = gst_sdp_message_get_attribute_val_n (sdp, "range", i);
+      if (range == NULL)
+        break;
+
+      /* keep track of the range and configure it in the segment */
+      if (gst_rtspsrc_parse_range (src, range, &src->segment))
+        break;
+    }
+  }
+  /* parse clock information. This is GStreamer specific, a server can tell the
+   * client what clock it is using and wrap that in a network clock. The
+   * advantage of that is that we can slave to it. */
+  {
+    const gchar *gstclock;
+
+    for (i = 0;; i++) {
+      gstclock = gst_sdp_message_get_attribute_val_n (sdp, "x-gst-clock", i);
+      if (gstclock == NULL)
+        break;
+
+      /* parse the clock and expose it in the provide_clock method */
+      if (gst_rtspsrc_parse_gst_clock (src, gstclock))
+        break;
+    }
+  }
+  /* try to find a global control attribute. Note that a '*' means that we should
+   * do aggregate control with the current url (so we don't do anything and
+   * leave the current connection as is) */
+  {
+    const gchar *control;
+
+    for (i = 0;; i++) {
+      control = gst_sdp_message_get_attribute_val_n (sdp, "control", i);
+      if (control == NULL)
+        break;
+
+      /* only take fully qualified urls */
+      if (g_str_has_prefix (control, "rtsp://"))
+        break;
+    }
+    if (control) {
+      g_free (src->conninfo.location);
+      src->conninfo.location = g_strdup (control);
+      /* make a connection for this, if there was a connection already, nothing
+       * happens. */
+      if (gst_rtsp_conninfo_connect (src, &src->conninfo, async) < 0) {
+        GST_ERROR_OBJECT (src, "could not connect");
+      }
+    }
+    /* we need to keep the control url separate from the connection url because
+     * the rules for constructing the media control url need it */
+    g_free (src->control);
+    src->control = g_strdup (control);
+  }
+
+  /* create streams */
+  n_streams = gst_sdp_message_medias_len (sdp);
+  for (i = 0; i < n_streams; i++) {
+    gst_rtspsrc_create_stream (src, sdp, i);
+  }
+
+  src->state = GST_RTSP_STATE_INIT;
+
+  /* setup streams */
+  if ((res = gst_rtspsrc_setup_streams (src, async)) < 0)
+    goto setup_failed;
+
+  /* reset our state */
+  src->need_range = TRUE;
+  src->skip = FALSE;
+
+  src->state = GST_RTSP_STATE_READY;
+
+  return res;
+
+  /* ERRORS */
+setup_failed:
+  {
+    GST_ERROR_OBJECT (src, "setup failed");
+    gst_rtspsrc_cleanup (src);
+    return res;
+  }
+}
+
+static GstRTSPResult
+gst_rtspsrc_retrieve_sdp (GstRTSPSrc * src, GstSDPMessage ** sdp,
+    gboolean async)
+{
+  GstRTSPResult res;
+  GstRTSPMessage request = { 0 };
+  GstRTSPMessage response = { 0 };
+  guint8 *data;
+  guint size;
+  gchar *respcont = NULL;
+
+restart:
+  src->need_redirect = FALSE;
+
+  /* can't continue without a valid url */
+  if (G_UNLIKELY (src->conninfo.url == NULL)) {
+    res = GST_RTSP_EINVAL;
+    goto no_url;
+  }
+  src->tried_url_auth = FALSE;
+
+  if ((res = gst_rtsp_conninfo_connect (src, &src->conninfo, async)) < 0)
+    goto connect_failed;
+
+  /* create OPTIONS */
+  GST_DEBUG_OBJECT (src, "create options...");
+  res =
+      gst_rtsp_message_init_request (&request, GST_RTSP_OPTIONS,
+      src->conninfo.url_str);
+  if (res < 0)
+    goto create_request_failed;
+
+  /* send OPTIONS */
+  GST_DEBUG_OBJECT (src, "send options...");
+
+  if (async)
+    GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving server options"));
+
+  if ((res =
+          gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
+              NULL)) < 0)
+    goto send_error;
+
+  /* parse OPTIONS */
+  if (!gst_rtspsrc_parse_methods (src, &response))
+    goto methods_error;
+
+  /* create DESCRIBE */
+  GST_DEBUG_OBJECT (src, "create describe...");
+  res =
+      gst_rtsp_message_init_request (&request, GST_RTSP_DESCRIBE,
+      src->conninfo.url_str);
+  if (res < 0)
+    goto create_request_failed;
+
+  /* we only accept SDP for now */
+  gst_rtsp_message_add_header (&request, GST_RTSP_HDR_ACCEPT,
+      "application/sdp");
+
+  /* send DESCRIBE */
+  GST_DEBUG_OBJECT (src, "send describe...");
+
+  if (async)
+    GST_ELEMENT_PROGRESS (src, CONTINUE, "open", ("Retrieving media info"));
+
+  if ((res =
+          gst_rtspsrc_send (src, src->conninfo.connection, &request, &response,
+              NULL)) < 0)
+    goto send_error;
+
+  /* we only perform redirect for the describe, currently */
+  if (src->need_redirect) {
+    /* close connection, we don't have to send a TEARDOWN yet, ignore the
+     * result. */
+    gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
+
+    gst_rtsp_message_unset (&request);
+    gst_rtsp_message_unset (&response);
+
+    /* and now retry */
+    goto restart;
+  }
+
+  /* it could be that the DESCRIBE method was not implemented */
+  if (!src->methods & GST_RTSP_DESCRIBE)
+    goto no_describe;
+
+  /* check if reply is SDP */
+  gst_rtsp_message_get_header (&response, GST_RTSP_HDR_CONTENT_TYPE, &respcont,
+      0);
+  /* could not be set but since the request returned OK, we assume it
+   * was SDP, else check it. */
+  if (respcont) {
+    if (g_ascii_strcasecmp (respcont, "application/sdp") != 0)
+      goto wrong_content_type;
+  }
+
+  /* get message body and parse as SDP */
+  gst_rtsp_message_get_body (&response, &data, &size);
+  if (data == NULL || size == 0)
+    goto no_describe;
+
+  GST_DEBUG_OBJECT (src, "parse SDP...");
+  gst_sdp_message_new (sdp);
+  gst_sdp_message_parse_buffer (data, size, *sdp);
+
+  /* clean up any messages */
+  gst_rtsp_message_unset (&request);
+  gst_rtsp_message_unset (&response);
+
+  return res;
+
+  /* ERRORS */
+no_url:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
+        ("No valid RTSP URL was provided"));
+    goto cleanup_error;
+  }
+connect_failed:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    if (res != GST_RTSP_EINTR) {
+      GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ_WRITE, (NULL),
+          ("Failed to connect. (%s)", str));
+    } else {
+      GST_WARNING_OBJECT (src, "connect interrupted");
+    }
+    g_free (str);
+    goto cleanup_error;
+  }
+create_request_failed:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
+        ("Could not create request. (%s)", str));
+    g_free (str);
+    goto cleanup_error;
+  }
+send_error:
+  {
+    /* Don't post a message - the rtsp_send method will have
+     * taken care of it because we passed NULL for the response code */
+    goto cleanup_error;
+  }
+methods_error:
+  {
+    /* error was posted */
+    res = GST_RTSP_ERROR;
+    goto cleanup_error;
+  }
+wrong_content_type:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
+        ("Server does not support SDP, got %s.", respcont));
+    res = GST_RTSP_ERROR;
+    goto cleanup_error;
+  }
+no_describe:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
+        ("Server can not provide an SDP."));
+    res = GST_RTSP_ERROR;
+    goto cleanup_error;
+  }
+cleanup_error:
+  {
+    if (src->conninfo.connection) {
+      GST_DEBUG_OBJECT (src, "free connection");
+      gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
+    }
+    gst_rtsp_message_unset (&request);
+    gst_rtsp_message_unset (&response);
+    return res;
+  }
+}
+
+static GstRTSPResult
+gst_rtspsrc_open (GstRTSPSrc * src, gboolean async)
+{
+  GstRTSPResult ret;
+
+  src->methods =
+      GST_RTSP_SETUP | GST_RTSP_PLAY | GST_RTSP_PAUSE | GST_RTSP_TEARDOWN;
+
+  if (src->sdp == NULL) {
+    if ((ret = gst_rtspsrc_retrieve_sdp (src, &src->sdp, async)) < 0)
+      goto no_sdp;
+  }
+
+  if ((ret = gst_rtspsrc_open_from_sdp (src, src->sdp, async)) < 0)
+    goto open_failed;
+
+done:
+  if (async)
+    gst_rtspsrc_loop_end_cmd (src, CMD_OPEN, ret);
+
+  return ret;
+
+  /* ERRORS */
+no_sdp:
+  {
+    GST_WARNING_OBJECT (src, "can't get sdp");
+    src->open_error = TRUE;
+    goto done;
+  }
+open_failed:
+  {
+    GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
+    src->open_error = TRUE;
+    goto done;
+  }
+}
+
+static GstRTSPResult
+gst_rtspsrc_close (GstRTSPSrc * src, gboolean async, gboolean only_close)
+{
+  GstRTSPMessage request = { 0 };
+  GstRTSPMessage response = { 0 };
+  GstRTSPResult res = GST_RTSP_OK;
+  GList *walk;
+  const gchar *control;
+
+  GST_DEBUG_OBJECT (src, "TEARDOWN...");
+
+  gst_rtspsrc_set_state (src, GST_STATE_READY);
+
+  if (src->state < GST_RTSP_STATE_READY) {
+    GST_DEBUG_OBJECT (src, "not ready, doing cleanup");
+    goto close;
+  }
+
+  if (only_close)
+    goto close;
+
+  /* construct a control url */
+  control = get_aggregate_control (src);
+
+  if (!(src->methods & (GST_RTSP_PLAY | GST_RTSP_TEARDOWN)))
+    goto not_supported;
+
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+    const gchar *setup_url;
+    GstRTSPConnInfo *info;
+
+    /* try aggregate control first but do non-aggregate control otherwise */
+    if (control)
+      setup_url = control;
+    else if ((setup_url = stream->conninfo.location) == NULL)
+      continue;
+
+    if (src->conninfo.connection) {
+      info = &src->conninfo;
+    } else if (stream->conninfo.connection) {
+      info = &stream->conninfo;
+    } else {
+      continue;
+    }
+    if (!info->connected)
+      goto next;
+
+    /* do TEARDOWN */
+    res =
+        gst_rtsp_message_init_request (&request, GST_RTSP_TEARDOWN, setup_url);
+    if (res < 0)
+      goto create_request_failed;
+
+    if (async)
+      GST_ELEMENT_PROGRESS (src, CONTINUE, "close", ("Closing stream"));
+
+    if ((res =
+            gst_rtspsrc_send (src, info->connection, &request, &response,
+                NULL)) < 0)
+      goto send_error;
+
+    /* FIXME, parse result? */
+    gst_rtsp_message_unset (&request);
+    gst_rtsp_message_unset (&response);
+
+  next:
+    /* early exit when we did aggregate control */
+    if (control)
+      break;
+  }
+
+close:
+  /* close connections */
+  GST_DEBUG_OBJECT (src, "closing connection...");
+  gst_rtsp_conninfo_close (src, &src->conninfo, TRUE);
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+    gst_rtsp_conninfo_close (src, &stream->conninfo, TRUE);
+  }
+
+  /* cleanup */
+  gst_rtspsrc_cleanup (src);
+
+  src->state = GST_RTSP_STATE_INVALID;
+
+  if (async)
+    gst_rtspsrc_loop_end_cmd (src, CMD_CLOSE, res);
+
+  return res;
+
+  /* ERRORS */
+create_request_failed:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
+        ("Could not create request. (%s)", str));
+    g_free (str);
+    goto close;
+  }
+send_error:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    gst_rtsp_message_unset (&request);
+    if (res != GST_RTSP_EINTR) {
+      GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
+          ("Could not send message. (%s)", str));
+    } else {
+      GST_WARNING_OBJECT (src, "TEARDOWN interrupted");
+    }
+    g_free (str);
+    goto close;
+  }
+not_supported:
+  {
+    GST_DEBUG_OBJECT (src,
+        "TEARDOWN and PLAY not supported, can't do TEARDOWN");
+    goto close;
+  }
+}
+
+/* RTP-Info is of the format:
+ *
+ * url=<URL>;[seq=<seqbase>;rtptime=<timebase>] [, url=...]
+ *
+ * rtptime corresponds to the timestamp for the NPT time given in the header
+ * seqbase corresponds to the next sequence number we received. This number
+ * indicates the first seqnum after the seek and should be used to discard
+ * packets that are from before the seek.
+ */
+static gboolean
+gst_rtspsrc_parse_rtpinfo (GstRTSPSrc * src, gchar * rtpinfo)
+{
+  gchar **infos;
+  gint i, j;
+
+  GST_DEBUG_OBJECT (src, "parsing RTP-Info %s", rtpinfo);
+
+  infos = g_strsplit (rtpinfo, ",", 0);
+  for (i = 0; infos[i]; i++) {
+    gchar **fields;
+    GstRTSPStream *stream;
+    gint32 seqbase;
+    gint64 timebase;
+
+    GST_DEBUG_OBJECT (src, "parsing info %s", infos[i]);
+
+    /* init values, types of seqbase and timebase are bigger than needed so we
+     * can store -1 as uninitialized values */
+    stream = NULL;
+    seqbase = -1;
+    timebase = -1;
+
+    /* parse url, find stream for url.
+     * parse seq and rtptime. The seq number should be configured in the rtp
+     * depayloader or session manager to detect gaps. Same for the rtptime, it
+     * should be used to create an initial time newsegment. */
+    fields = g_strsplit (infos[i], ";", 0);
+    for (j = 0; fields[j]; j++) {
+      GST_DEBUG_OBJECT (src, "parsing field %s", fields[j]);
+      /* remove leading whitespace */
+      fields[j] = g_strchug (fields[j]);
+      if (g_str_has_prefix (fields[j], "url=")) {
+        /* get the url and the stream */
+        stream =
+            find_stream (src, (fields[j] + 4), (gpointer) find_stream_by_setup);
+      } else if (g_str_has_prefix (fields[j], "seq=")) {
+        seqbase = atoi (fields[j] + 4);
+      } else if (g_str_has_prefix (fields[j], "rtptime=")) {
+        timebase = g_ascii_strtoll (fields[j] + 8, NULL, 10);
+      }
+    }
+    g_strfreev (fields);
+    /* now we need to store the values for the caps of the stream */
+    if (stream != NULL) {
+      GST_DEBUG_OBJECT (src,
+          "found stream %p, setting: seqbase %d, timebase %" G_GINT64_FORMAT,
+          stream, seqbase, timebase);
+
+      /* we have a stream, configure detected params */
+      stream->seqbase = seqbase;
+      stream->timebase = timebase;
+    }
+  }
+  g_strfreev (infos);
+
+  return TRUE;
+}
+
+static void
+gst_rtspsrc_handle_rtcp_interval (GstRTSPSrc * src, gchar * rtcp)
+{
+  guint64 interval;
+  GList *walk;
+
+  interval = strtoul (rtcp, NULL, 10);
+  GST_DEBUG_OBJECT (src, "rtcp interval: %" G_GUINT64_FORMAT " ms", interval);
+
+  if (!interval)
+    return;
+
+  interval *= GST_MSECOND;
+
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+
+    /* already (optionally) retrieved this when configuring manager */
+    if (stream->session) {
+      GObject *rtpsession = stream->session;
+
+      GST_DEBUG_OBJECT (src, "configure rtcp interval in session %p",
+          rtpsession);
+      g_object_set (rtpsession, "rtcp-min-interval", interval, NULL);
+    }
+  }
+
+  /* now it happens that (Xenon) server sending this may also provide bogus
+   * RTCP SR sync data (i.e. with quite some jitter), so never mind those
+   * and just use RTP-Info to sync */
+  if (src->manager) {
+    GObjectClass *klass;
+
+    klass = G_OBJECT_GET_CLASS (G_OBJECT (src->manager));
+    if (g_object_class_find_property (klass, "rtcp-sync")) {
+      GST_DEBUG_OBJECT (src, "configuring rtp sync method");
+      g_object_set (src->manager, "rtcp-sync", RTCP_SYNC_RTP, NULL);
+    }
+  }
+}
+
+static gdouble
+gst_rtspsrc_get_float (const gchar * dstr)
+{
+  gchar s[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
+
+  /* canonicalise floating point string so we can handle float strings
+   * in the form "24.930" or "24,930" irrespective of the current locale */
+  g_strlcpy (s, dstr, sizeof (s));
+  g_strdelimit (s, ",", '.');
+  return g_ascii_strtod (s, NULL);
+}
+
+static gchar *
+gen_range_header (GstRTSPSrc * src, GstSegment * segment)
+{
+  gchar val_str[G_ASCII_DTOSTR_BUF_SIZE] = { 0, };
+#ifdef GST_EXT_RTSP_MODIFICATION
+  if (src->start_position !=0 && segment->position == 0) {
+    segment->position = src->start_position;
+    src->start_position = 0;
+  }
+#endif
+  if (src->range && src->range->min.type == GST_RTSP_TIME_NOW) {
+    g_strlcpy (val_str, "now", sizeof (val_str));
+  } else {
+    if (segment->position == 0) {
+      g_strlcpy (val_str, "0", sizeof (val_str));
+    } else {
+      g_ascii_dtostr (val_str, sizeof (val_str),
+          ((gdouble) segment->position) / GST_SECOND);
+    }
+  }
+  return g_strdup_printf ("npt=%s-", val_str);
+}
+
+static void
+clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
+{
+  guint i, len;
+
+  stream->timebase = -1;
+  stream->seqbase = -1;
+
+  len = stream->ptmap->len;
+  for (i = 0; i < len; i++) {
+    PtMapItem *item = &g_array_index (stream->ptmap, PtMapItem, i);
+    GstStructure *s;
+
+    if (item->caps == NULL)
+      continue;
+
+    item->caps = gst_caps_make_writable (item->caps);
+    s = gst_caps_get_structure (item->caps, 0);
+    gst_structure_remove_fields (s, "clock-base", "seqnum-base", NULL);
+  }
+}
+
+static GstRTSPResult
+gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+
+  if (src->state < GST_RTSP_STATE_READY) {
+    res = GST_RTSP_ERROR;
+    if (src->open_error) {
+      GST_DEBUG_OBJECT (src, "the stream was in error");
+      goto done;
+    }
+    if (async)
+      gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
+
+    if ((res = gst_rtspsrc_open (src, async)) < 0) {
+      GST_DEBUG_OBJECT (src, "failed to open stream");
+      goto done;
+    }
+  }
+
+done:
+  return res;
+}
+
+static GstRTSPResult
+gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
+{
+  GstRTSPMessage request = { 0 };
+  GstRTSPMessage response = { 0 };
+  GstRTSPResult res = GST_RTSP_OK;
+  GList *walk;
+  gchar *hval;
+  gint hval_idx;
+  const gchar *control;
+
+  GST_DEBUG_OBJECT (src, "PLAY...");
+
+  if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
+    goto open_failed;
+
+  if (!(src->methods & GST_RTSP_PLAY))
+    goto not_supported;
+
+  if (src->state == GST_RTSP_STATE_PLAYING)
+    goto was_playing;
+
+  if (!src->conninfo.connection || !src->conninfo.connected)
+    goto done;
+
+  /* send some dummy packets before we activate the receive in the
+   * udp sources */
+  gst_rtspsrc_send_dummy_packets (src);
+
+  /* require new SR packets */
+  if (src->manager)
+    g_signal_emit_by_name (src->manager, "reset-sync", NULL);
+
+  /* construct a control url */
+  control = get_aggregate_control (src);
+
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+    const gchar *setup_url;
+    GstRTSPConnection *conn;
+
+    /* try aggregate control first but do non-aggregate control otherwise */
+    if (control)
+      setup_url = control;
+    else if ((setup_url = stream->conninfo.location) == NULL)
+      continue;
+
+    if (src->conninfo.connection) {
+      conn = src->conninfo.connection;
+    } else if (stream->conninfo.connection) {
+      conn = stream->conninfo.connection;
+    } else {
+      continue;
+    }
+
+    /* do play */
+    res = gst_rtsp_message_init_request (&request, GST_RTSP_PLAY, setup_url);
+    if (res < 0)
+      goto create_request_failed;
+
+    if (src->need_range) {
+      hval = gen_range_header (src, segment);
+
+      gst_rtsp_message_take_header (&request, GST_RTSP_HDR_RANGE, hval);
+
+      /* store the newsegment event so it can be sent from the streaming thread. */
+      src->need_segment = TRUE;
+    }
+
+    if (segment->rate != 1.0) {
+      gchar hval[G_ASCII_DTOSTR_BUF_SIZE];
+
+      g_ascii_dtostr (hval, sizeof (hval), segment->rate);
+      if (src->skip)
+        gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SCALE, hval);
+      else
+        gst_rtsp_message_add_header (&request, GST_RTSP_HDR_SPEED, hval);
+    }
+
+    if (async)
+      GST_ELEMENT_PROGRESS (src, CONTINUE, "request", ("Sending PLAY request"));
+
+    if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
+      goto send_error;
+
+    /* seek may have silently failed as it is not supported */
+    if (!(src->methods & GST_RTSP_PLAY)) {
+      GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");
+      /* obviously it is supported as we made it here */
+      src->methods |= GST_RTSP_PLAY;
+      src->seekable = FALSE;
+      /* but there is nothing to parse in the response,
+       * so convey we have no idea and not to expect anything particular */
+      clear_rtp_base (src, stream);
+      if (control) {
+        GList *run;
+
+        /* need to do for all streams */
+        for (run = src->streams; run; run = g_list_next (run))
+          clear_rtp_base (src, (GstRTSPStream *) run->data);
+      }
+      /* NOTE the above also disables npt based eos detection */
+      /* and below forces position to 0,
+       * which is visible feedback we lost the plot */
+      segment->start = segment->position = src->last_pos;
+    }
+
+    gst_rtsp_message_unset (&request);
+
+    /* parse RTP npt field. This is the current position in the stream (Normal
+     * Play Time) and should be put in the NEWSEGMENT position field. */
+    if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RANGE, &hval,
+            0) == GST_RTSP_OK)
+      gst_rtspsrc_parse_range (src, hval, segment);
+
+    /* assume 1.0 rate now, overwrite when the SCALE or SPEED headers are present. */
+    segment->rate = 1.0;
+
+    /* parse Speed header. This is the intended playback rate of the stream
+     * and should be put in the NEWSEGMENT rate field. */
+    if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SPEED, &hval,
+            0) == GST_RTSP_OK) {
+      segment->rate = gst_rtspsrc_get_float (hval);
+    } else if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_SCALE,
+            &hval, 0) == GST_RTSP_OK) {
+      segment->rate = gst_rtspsrc_get_float (hval);
+    }
+
+    /* parse the RTP-Info header field (if ANY) to get the base seqnum and timestamp
+     * for the RTP packets. If this is not present, we assume all starts from 0...
+     * This is info for the RTP session manager that we pass to it in caps. */
+    hval_idx = 0;
+    while (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTP_INFO,
+            &hval, hval_idx++) == GST_RTSP_OK)
+      gst_rtspsrc_parse_rtpinfo (src, hval);
+
+    /* some servers indicate RTCP parameters in PLAY response,
+     * rather than properly in SDP */
+    if (gst_rtsp_message_get_header (&response, GST_RTSP_HDR_RTCP_INTERVAL,
+            &hval, 0) == GST_RTSP_OK)
+      gst_rtspsrc_handle_rtcp_interval (src, hval);
+
+    gst_rtsp_message_unset (&response);
+
+    /* early exit when we did aggregate control */
+    if (control)
+      break;
+  }
+  /* configure the caps of the streams after we parsed all headers. Only reset
+   * the manager object when we set a new Range header (we did a seek) */
+  gst_rtspsrc_configure_caps (src, segment, src->need_range);
+
+  /* set to PLAYING after we have configured the caps, otherwise we
+   * might end up calling request_key (with SRTP) while caps are still
+   * being configured. */
+  gst_rtspsrc_set_state (src, GST_STATE_PLAYING);
+
+  /* set again when needed */
+  src->need_range = FALSE;
+
+  src->running = TRUE;
+  src->base_time = -1;
+  src->state = GST_RTSP_STATE_PLAYING;
+
+  /* mark discont */
+  GST_DEBUG_OBJECT (src, "mark DISCONT, we did a seek to another position");
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+    stream->discont = TRUE;
+  }
+
+done:
+  if (async)
+    gst_rtspsrc_loop_end_cmd (src, CMD_PLAY, res);
+
+  return res;
+
+  /* ERRORS */
+open_failed:
+  {
+    GST_DEBUG_OBJECT (src, "failed to open stream");
+    goto done;
+  }
+not_supported:
+  {
+    GST_DEBUG_OBJECT (src, "PLAY is not supported");
+    goto done;
+  }
+was_playing:
+  {
+    GST_DEBUG_OBJECT (src, "we were already PLAYING");
+    goto done;
+  }
+create_request_failed:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
+        ("Could not create request. (%s)", str));
+    g_free (str);
+    goto done;
+  }
+send_error:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    gst_rtsp_message_unset (&request);
+    if (res != GST_RTSP_EINTR) {
+      GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
+          ("Could not send message. (%s)", str));
+    } else {
+      GST_WARNING_OBJECT (src, "PLAY interrupted");
+    }
+    g_free (str);
+    goto done;
+  }
+}
+
+static GstRTSPResult
+gst_rtspsrc_pause (GstRTSPSrc * src, gboolean async)
+{
+  GstRTSPResult res = GST_RTSP_OK;
+  GstRTSPMessage request = { 0 };
+  GstRTSPMessage response = { 0 };
+  GList *walk;
+  const gchar *control;
+
+  GST_DEBUG_OBJECT (src, "PAUSE...");
+
+  if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
+    goto open_failed;
+
+  if (!(src->methods & GST_RTSP_PAUSE))
+    goto not_supported;
+
+  if (src->state == GST_RTSP_STATE_READY)
+    goto was_paused;
+
+  if (!src->conninfo.connection || !src->conninfo.connected)
+    goto no_connection;
+
+  /* construct a control url */
+  control = get_aggregate_control (src);
+
+  /* loop over the streams. We might exit the loop early when we could do an
+   * aggregate control */
+  for (walk = src->streams; walk; walk = g_list_next (walk)) {
+    GstRTSPStream *stream = (GstRTSPStream *) walk->data;
+    GstRTSPConnection *conn;
+    const gchar *setup_url;
+
+    /* try aggregate control first but do non-aggregate control otherwise */
+    if (control)
+      setup_url = control;
+    else if ((setup_url = stream->conninfo.location) == NULL)
+      continue;
+
+    if (src->conninfo.connection) {
+      conn = src->conninfo.connection;
+    } else if (stream->conninfo.connection) {
+      conn = stream->conninfo.connection;
+    } else {
+      continue;
+    }
+
+    if (async)
+      GST_ELEMENT_PROGRESS (src, CONTINUE, "request",
+          ("Sending PAUSE request"));
+
+    if ((res =
+            gst_rtsp_message_init_request (&request, GST_RTSP_PAUSE,
+                setup_url)) < 0)
+      goto create_request_failed;
+
+    if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
+      goto send_error;
+
+    gst_rtsp_message_unset (&request);
+    gst_rtsp_message_unset (&response);
+
+    /* exit early when we did agregate control */
+    if (control)
+      break;
+  }
+
+  /* change element states now */
+  gst_rtspsrc_set_state (src, GST_STATE_PAUSED);
+
+no_connection:
+  src->state = GST_RTSP_STATE_READY;
+
+done:
+  if (async)
+    gst_rtspsrc_loop_end_cmd (src, CMD_PAUSE, res);
+
+  return res;
+
+  /* ERRORS */
+open_failed:
+  {
+    GST_DEBUG_OBJECT (src, "failed to open stream");
+    goto done;
+  }
+not_supported:
+  {
+    GST_DEBUG_OBJECT (src, "PAUSE is not supported");
+    goto done;
+  }
+was_paused:
+  {
+    GST_DEBUG_OBJECT (src, "we were already PAUSED");
+    goto done;
+  }
+create_request_failed:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    GST_ELEMENT_ERROR (src, LIBRARY, INIT, (NULL),
+        ("Could not create request. (%s)", str));
+    g_free (str);
+    goto done;
+  }
+send_error:
+  {
+    gchar *str = gst_rtsp_strresult (res);
+
+    gst_rtsp_message_unset (&request);
+    if (res != GST_RTSP_EINTR) {
+      GST_ELEMENT_ERROR (src, RESOURCE, WRITE, (NULL),
+          ("Could not send message. (%s)", str));
+    } else {
+      GST_WARNING_OBJECT (src, "PAUSE interrupted");
+    }
+    g_free (str);
+    goto done;
+  }
+}
+
+static void
+gst_rtspsrc_handle_message (GstBin * bin, GstMessage * message)
+{
+  GstRTSPSrc *rtspsrc;
+
+  rtspsrc = GST_RTSPSRC (bin);
+
+  switch (GST_MESSAGE_TYPE (message)) {
+    case GST_MESSAGE_EOS:
+      gst_message_unref (message);
+      break;
+    case GST_MESSAGE_ELEMENT:
+    {
+      const GstStructure *s = gst_message_get_structure (message);
+
+      if (gst_structure_has_name (s, "GstUDPSrcTimeout")) {
+        gboolean ignore_timeout;
+
+        GST_DEBUG_OBJECT (bin, "timeout on UDP port");
+
+        GST_OBJECT_LOCK (rtspsrc);
+        ignore_timeout = rtspsrc->ignore_timeout;
+        rtspsrc->ignore_timeout = TRUE;
+        GST_OBJECT_UNLOCK (rtspsrc);
+
+        /* we only act on the first udp timeout message, others are irrelevant
+         * and can be ignored. */
+        if (!ignore_timeout)
+          gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_RECONNECT, CMD_LOOP);
+        /* eat and free */
+        gst_message_unref (message);
+        return;
+      }
+      GST_BIN_CLASS (parent_class)->handle_message (bin, message);
+      break;
+    }
+    case GST_MESSAGE_ERROR:
+    {
+      GstObject *udpsrc;
+      GstRTSPStream *stream;
+      GstFlowReturn ret;
+
+      udpsrc = GST_MESSAGE_SRC (message);
+
+      GST_DEBUG_OBJECT (rtspsrc, "got error from %s",
+          GST_ELEMENT_NAME (udpsrc));
+
+      stream = find_stream (rtspsrc, udpsrc, (gpointer) find_stream_by_udpsrc);
+      if (!stream)
+        goto forward;
+
+      /* we ignore the RTCP udpsrc */
+      if (stream->udpsrc[1] == GST_ELEMENT_CAST (udpsrc))
+        goto done;
+
+      /* if we get error messages from the udp sources, that's not a problem as
+       * long as not all of them error out. We also don't really know what the
+       * problem is, the message does not give enough detail... */
+      ret = gst_rtspsrc_combine_flows (rtspsrc, stream, GST_FLOW_NOT_LINKED);
+      GST_DEBUG_OBJECT (rtspsrc, "combined flows: %s", gst_flow_get_name (ret));
+      if (ret != GST_FLOW_OK)
+        goto forward;
+
+    done:
+      gst_message_unref (message);
+      break;
+
+    forward:
+      /* fatal but not our message, forward */
+      GST_BIN_CLASS (parent_class)->handle_message (bin, message);
+      break;
+    }
+    default:
+    {
+      GST_BIN_CLASS (parent_class)->handle_message (bin, message);
+      break;
+    }
+  }
+}
+
+/* the thread where everything happens */
+static void
+gst_rtspsrc_thread (GstRTSPSrc * src)
+{
+  gint cmd;
+
+  GST_OBJECT_LOCK (src);
+  cmd = src->pending_cmd;
+  if (cmd == CMD_RECONNECT || cmd == CMD_PLAY || cmd == CMD_PAUSE
+      || cmd == CMD_LOOP || cmd == CMD_OPEN)
+    src->pending_cmd = CMD_LOOP;
+  else
+    src->pending_cmd = CMD_WAIT;
+  GST_DEBUG_OBJECT (src, "got command %s", cmd_to_string (cmd));
+
+  /* we got the message command, so ensure communication is possible again */
+  gst_rtspsrc_connection_flush (src, FALSE);
+
+  src->busy_cmd = cmd;
+  GST_OBJECT_UNLOCK (src);
+
+  switch (cmd) {
+    case CMD_OPEN:
+      gst_rtspsrc_open (src, TRUE);
+      break;
+    case CMD_PLAY:
+      gst_rtspsrc_play (src, &src->segment, TRUE);
+      break;
+    case CMD_PAUSE:
+      gst_rtspsrc_pause (src, TRUE);
+      break;
+    case CMD_CLOSE:
+      gst_rtspsrc_close (src, TRUE, FALSE);
+      break;
+    case CMD_LOOP:
+      gst_rtspsrc_loop (src);
+      break;
+    case CMD_RECONNECT:
+      gst_rtspsrc_reconnect (src, FALSE);
+      break;
+    default:
+      break;
+  }
+
+  GST_OBJECT_LOCK (src);
+  /* and go back to sleep */
+  if (src->pending_cmd == CMD_WAIT) {
+    if (src->task)
+      gst_task_pause (src->task);
+  }
+  /* reset waiting */
+  src->busy_cmd = CMD_WAIT;
+  GST_OBJECT_UNLOCK (src);
+}
+
+static gboolean
+gst_rtspsrc_start (GstRTSPSrc * src)
+{
+  GST_DEBUG_OBJECT (src, "starting");
+
+  GST_OBJECT_LOCK (src);
+
+  src->pending_cmd = CMD_WAIT;
+
+  if (src->task == NULL) {
+    src->task = gst_task_new ((GstTaskFunction) gst_rtspsrc_thread, src, NULL);
+    if (src->task == NULL)
+      goto task_error;
+
+    gst_task_set_lock (src->task, GST_RTSP_STREAM_GET_LOCK (src));
+  }
+  GST_OBJECT_UNLOCK (src);
+
+  return TRUE;
+
+  /* ERRORS */
+task_error:
+  {
+    GST_OBJECT_UNLOCK (src);
+    GST_ERROR_OBJECT (src, "failed to create task");
+    return FALSE;
+  }
+}
+
+static gboolean
+gst_rtspsrc_stop (GstRTSPSrc * src)
+{
+  GstTask *task;
+
+  GST_DEBUG_OBJECT (src, "stopping");
+
+  /* also cancels pending task */
+  gst_rtspsrc_loop_send_cmd (src, CMD_WAIT, CMD_ALL);
+
+  GST_OBJECT_LOCK (src);
+  if ((task = src->task)) {
+    src->task = NULL;
+    GST_OBJECT_UNLOCK (src);
+
+    gst_task_stop (task);
+
+    /* make sure it is not running */
+    GST_RTSP_STREAM_LOCK (src);
+    GST_RTSP_STREAM_UNLOCK (src);
+
+    /* now wait for the task to finish */
+    gst_task_join (task);
+
+    /* and free the task */
+    gst_object_unref (GST_OBJECT (task));
+
+    GST_OBJECT_LOCK (src);
+  }
+  GST_OBJECT_UNLOCK (src);
+
+  /* ensure synchronously all is closed and clean */
+  gst_rtspsrc_close (src, FALSE, TRUE);
+
+  return TRUE;
+}
+
+static GstStateChangeReturn
+gst_rtspsrc_change_state (GstElement * element, GstStateChange transition)
+{
+  GstRTSPSrc *rtspsrc;
+  GstStateChangeReturn ret;
+#ifdef GST_EXT_RTSP_MODIFICATION
+  guint64 end_time;
+#endif
+
+  rtspsrc = GST_RTSPSRC (element);
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      if (!gst_rtspsrc_start (rtspsrc))
+        goto start_failed;
+      break;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+      /* init some state */
+      rtspsrc->cur_protocols = rtspsrc->protocols;
+      /* first attempt, don't ignore timeouts */
+      rtspsrc->ignore_timeout = FALSE;
+      rtspsrc->open_error = FALSE;
+      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_OPEN, 0);
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+      set_manager_buffer_mode (rtspsrc);
+      /* fall-through */
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      /* unblock the tcp tasks and make the loop waiting */
+      if (gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_WAIT, CMD_LOOP)) {
+        /* make sure it is waiting before we send PAUSE or PLAY below */
+        GST_RTSP_STREAM_LOCK (rtspsrc);
+        GST_RTSP_STREAM_UNLOCK (rtspsrc);
+      }
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      break;
+    default:
+      break;
+  }
+
+  ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+  if (ret == GST_STATE_CHANGE_FAILURE)
+    goto done;
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+      ret = GST_STATE_CHANGE_SUCCESS;
+      break;
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+#ifdef GST_EXT_RTSP_MODIFICATION
+      /* don't change to PAUSE state before complete stream opend.
+              see gst_rtspsrc_loop_complete_cmd() */
+      g_mutex_lock(&(rtspsrc)->pause_lock);
+      end_time = g_get_monotonic_time () + 10 * G_TIME_SPAN_SECOND;
+      if (!g_cond_wait_until (&(rtspsrc)->open_end, &(rtspsrc)->pause_lock, end_time)) {
+        GST_WARNING_OBJECT(rtspsrc, "time out: stream opend is not completed yet..");
+      }
+      g_mutex_unlock(&(rtspsrc)->pause_lock);
+#endif
+      ret = GST_STATE_CHANGE_NO_PREROLL;
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
+      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PLAY, 0);
+      ret = GST_STATE_CHANGE_SUCCESS;
+      break;
+    case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
+      /* send pause request and keep the idle task around */
+      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_PAUSE, CMD_LOOP);
+      ret = GST_STATE_CHANGE_NO_PREROLL;
+      break;
+    case GST_STATE_CHANGE_PAUSED_TO_READY:
+      gst_rtspsrc_loop_send_cmd (rtspsrc, CMD_CLOSE, CMD_PAUSE);
+      ret = GST_STATE_CHANGE_SUCCESS;
+      break;
+    case GST_STATE_CHANGE_READY_TO_NULL:
+      gst_rtspsrc_stop (rtspsrc);
+      ret = GST_STATE_CHANGE_SUCCESS;
+      break;
+    default:
+      break;
+  }
+
+done:
+  return ret;
+
+start_failed:
+  {
+    GST_DEBUG_OBJECT (rtspsrc, "start failed");
+    return GST_STATE_CHANGE_FAILURE;
+  }
+}
+
+static gboolean
+gst_rtspsrc_send_event (GstElement * element, GstEvent * event)
+{
+  gboolean res;
+  GstRTSPSrc *rtspsrc;
+
+  rtspsrc = GST_RTSPSRC (element);
+
+  if (GST_EVENT_IS_DOWNSTREAM (event)) {
+    res = gst_rtspsrc_push_event (rtspsrc, event);
+  } else {
+    res = GST_ELEMENT_CLASS (parent_class)->send_event (element, event);
+  }
+
+  return res;
+}
+
+
+/*** GSTURIHANDLER INTERFACE *************************************************/
+
+static GstURIType
+gst_rtspsrc_uri_get_type (GType type)
+{
+  return GST_URI_SRC;
+}
+
+static const gchar *const *
+gst_rtspsrc_uri_get_protocols (GType type)
+{
+  static const gchar *protocols[] =
+      { "rtsp", "rtspu", "rtspt", "rtsph", "rtsp-sdp",
+    "rtsps", "rtspsu", "rtspst", "rtspsh", NULL
+  };
+
+  return protocols;
+}
+
+static gchar *
+gst_rtspsrc_uri_get_uri (GstURIHandler * handler)
+{
+  GstRTSPSrc *src = GST_RTSPSRC (handler);
+
+  /* FIXME: make thread-safe */
+  return g_strdup (src->conninfo.location);
+}
+
+static gboolean
+gst_rtspsrc_uri_set_uri (GstURIHandler * handler, const gchar * uri,
+    GError ** error)
+{
+  GstRTSPSrc *src;
+  GstRTSPResult res;
+  GstSDPResult sres;
+  GstRTSPUrl *newurl = NULL;
+  GstSDPMessage *sdp = NULL;
+
+  src = GST_RTSPSRC (handler);
+
+  /* same URI, we're fine */
+  if (src->conninfo.location && uri && !strcmp (uri, src->conninfo.location))
+    goto was_ok;
+
+  if (g_str_has_prefix (uri, "rtsp-sdp://")) {
+    sres = gst_sdp_message_new (&sdp);
+    if (sres < 0)
+      goto sdp_failed;
+
+    GST_DEBUG_OBJECT (src, "parsing SDP message");
+    sres = gst_sdp_message_parse_uri (uri, sdp);
+    if (sres < 0)
+      goto invalid_sdp;
+  } else {
+    /* try to parse */
+    GST_DEBUG_OBJECT (src, "parsing URI");
+    if ((res = gst_rtsp_url_parse (uri, &newurl)) < 0)
+      goto parse_error;
+  }
+
+  /* if worked, free previous and store new url object along with the original
+   * location. */
+  GST_DEBUG_OBJECT (src, "configuring URI");
+  g_free (src->conninfo.location);
+  src->conninfo.location = g_strdup (uri);
+  gst_rtsp_url_free (src->conninfo.url);
+  src->conninfo.url = newurl;
+  g_free (src->conninfo.url_str);
+  if (newurl)
+    src->conninfo.url_str = gst_rtsp_url_get_request_uri (src->conninfo.url);
+  else
+    src->conninfo.url_str = NULL;
+
+  if (src->sdp)
+    gst_sdp_message_free (src->sdp);
+  src->sdp = sdp;
+  src->from_sdp = sdp != NULL;
+
+  GST_DEBUG_OBJECT (src, "set uri: %s", GST_STR_NULL (uri));
+  GST_DEBUG_OBJECT (src, "request uri is: %s",
+      GST_STR_NULL (src->conninfo.url_str));
+
+  return TRUE;
+
+  /* Special cases */
+was_ok:
+  {
+    GST_DEBUG_OBJECT (src, "URI was ok: '%s'", GST_STR_NULL (uri));
+    return TRUE;
+  }
+sdp_failed:
+  {
+    GST_ERROR_OBJECT (src, "Could not create new SDP (%d)", sres);
+    g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
+        "Could not create SDP");
+    return FALSE;
+  }
+invalid_sdp:
+  {
+    GST_ERROR_OBJECT (src, "Not a valid SDP (%d) '%s'", sres,
+        GST_STR_NULL (uri));
+    gst_sdp_message_free (sdp);
+    g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
+        "Invalid SDP");
+    return FALSE;
+  }
+parse_error:
+  {
+    GST_ERROR_OBJECT (src, "Not a valid RTSP url '%s' (%d)",
+        GST_STR_NULL (uri), res);
+    g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
+        "Invalid RTSP URI");
+    return FALSE;
+  }
+}
+
+static void
+gst_rtspsrc_uri_handler_init (gpointer g_iface, gpointer iface_data)
+{
+  GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
+
+  iface->get_type = gst_rtspsrc_uri_get_type;
+  iface->get_protocols = gst_rtspsrc_uri_get_protocols;
+  iface->get_uri = gst_rtspsrc_uri_get_uri;
+  iface->set_uri = gst_rtspsrc_uri_set_uri;
+}
diff --git a/gst/rtsp/gstrtspsrc.h b/gst/rtsp/gstrtspsrc.h
new file mode 100755
index 0000000..6309aeb
--- /dev/null
+++ b/gst/rtsp/gstrtspsrc.h
@@ -0,0 +1,283 @@
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ *               <2006> Wim Taymans <wim@fluendo.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+/*
+ * Unless otherwise indicated, Source Code is licensed under MIT license.
+ * See further explanation attached in License Statement (distributed in the file
+ * LICENSE).
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a copy of
+ * this software and associated documentation files (the "Software"), to deal in
+ * the Software without restriction, including without limitation the rights to
+ * use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies
+ * of the Software, and to permit persons to whom the Software is furnished to do
+ * so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in all
+ * copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM,
+ * OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE
+ * SOFTWARE.
+ */
+
+#ifndef __GST_RTSPSRC_H__
+#define __GST_RTSPSRC_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+#include <gst/rtsp/rtsp.h>
+#include <gio/gio.h>
+
+#include "gstrtspext.h"
+
+#define GST_TYPE_RTSPSRC \
+  (gst_rtspsrc_get_type())
+#define GST_RTSPSRC(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTSPSRC,GstRTSPSrc))
+#define GST_RTSPSRC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTSPSRC,GstRTSPSrcClass))
+#define GST_IS_RTSPSRC(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTSPSRC))
+#define GST_IS_RTSPSRC_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTSPSRC))
+#define GST_RTSPSRC_CAST(obj) \
+  ((GstRTSPSrc *)(obj))
+
+typedef struct _GstRTSPSrc GstRTSPSrc;
+typedef struct _GstRTSPSrcClass GstRTSPSrcClass;
+
+#define GST_RTSP_STATE_GET_LOCK(rtsp)    (&GST_RTSPSRC_CAST(rtsp)->state_rec_lock)
+#define GST_RTSP_STATE_LOCK(rtsp)        (g_rec_mutex_lock (GST_RTSP_STATE_GET_LOCK(rtsp)))
+#define GST_RTSP_STATE_UNLOCK(rtsp)      (g_rec_mutex_unlock (GST_RTSP_STATE_GET_LOCK(rtsp)))
+
+#define GST_RTSP_STREAM_GET_LOCK(rtsp)   (&GST_RTSPSRC_CAST(rtsp)->stream_rec_lock)
+#define GST_RTSP_STREAM_LOCK(rtsp)       (g_rec_mutex_lock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
+#define GST_RTSP_STREAM_UNLOCK(rtsp)     (g_rec_mutex_unlock (GST_RTSP_STREAM_GET_LOCK(rtsp)))
+
+typedef struct _GstRTSPConnInfo GstRTSPConnInfo;
+
+struct _GstRTSPConnInfo {
+  gchar              *location;
+  GstRTSPUrl         *url;
+  gchar              *url_str;
+  GstRTSPConnection  *connection;
+  gboolean            connected;
+  gboolean            flushing;
+};
+
+typedef struct _GstRTSPStream GstRTSPStream;
+
+struct _GstRTSPStream {
+  gint          id;
+
+  GstRTSPSrc   *parent; /* parent, no extra ref to parent is taken */
+
+  /* pad we expose or NULL when it does not have an actual pad */
+  GstPad       *srcpad;
+  GstFlowReturn last_ret;
+  gboolean      added;
+  gboolean      setup;
+  gboolean      skipped;
+  gboolean      eos;
+  gboolean      discont;
+
+  /* for interleaved mode */
+  guint8        channel[2];
+  GstPad       *channelpad[2];
+
+  /* our udp sources */
+  GstElement   *udpsrc[2];
+  GstPad       *blockedpad;
+  gulong        blockid;
+  gboolean      is_ipv6;
+
+  /* our udp sinks back to the server */
+  GstElement   *udpsink[2];
+  GstPad       *rtcppad;
+
+  /* fakesrc for sending dummy data */
+  GstElement   *fakesrc;
+
+  /* state */
+  guint         port;
+  gboolean      container;
+  gboolean      is_real;
+  guint8        default_pt;
+  GstRTSPProfile profile;
+  GArray       *ptmap;
+  /* original control url */
+  gchar        *control_url;
+  guint32       ssrc;
+  guint32       seqbase;
+  guint64       timebase;
+  GstElement   *srtpdec;
+  GstCaps      *srtcpparams;
+  GstElement   *srtpenc;
+  guint32       send_ssrc;
+
+  /* per stream connection */
+  GstRTSPConnInfo  conninfo;
+
+  /* session */
+  GObject      *session;
+
+  /* bandwidth */
+  guint         as_bandwidth;
+  guint         rs_bandwidth;
+  guint         rr_bandwidth;
+
+  /* destination */
+  gchar        *destination;
+  gboolean      is_multicast;
+  guint         ttl;
+
+  GstStructure     *rtx_pt_map;
+};
+
+/**
+ * GstRTSPNatMethod:
+ * @GST_RTSP_NAT_NONE: none
+ * @GST_RTSP_NAT_DUMMY: send dummy packets
+ *
+ * Different methods for trying to traverse firewalls.
+ */
+typedef enum
+{
+  GST_RTSP_NAT_NONE,
+  GST_RTSP_NAT_DUMMY
+} GstRTSPNatMethod;
+
+struct _GstRTSPSrc {
+  GstBin           parent;
+
+  /* task and mutex for interleaved mode */
+  gboolean         interleaved;
+  GstTask         *task;
+  GRecMutex        stream_rec_lock;
+  GstSegment       segment;
+  gboolean         running;
+  gboolean         need_range;
+  gboolean         skip;
+  gint             free_channel;
+  gboolean         need_segment;
+  GstClockTime     base_time;
+
+  /* UDP mode loop */
+  gint             pending_cmd;
+  gint             busy_cmd;
+  gboolean         ignore_timeout;
+  gboolean         open_error;
+
+  /* mutex for protecting state changes */
+  GRecMutex        state_rec_lock;
+
+  GstSDPMessage   *sdp;
+  gboolean         from_sdp;
+  GList           *streams;
+  GstStructure    *props;
+  gboolean         need_activate;
+
+  /* properties */
+  GstRTSPLowerTrans protocols;
+  gboolean          debug;
+  guint             retry;
+  guint64           udp_timeout;
+  GTimeVal          tcp_timeout;
+  GTimeVal         *ptcp_timeout;
+  guint             latency;
+  gboolean          drop_on_latency;
+  guint64           connection_speed;
+  GstRTSPNatMethod  nat_method;
+  gboolean          do_rtcp;
+  gboolean          do_rtsp_keep_alive;
+  gchar            *proxy_host;
+  guint             proxy_port;
+  gchar            *proxy_user;        /* from url or property */
+  gchar            *proxy_passwd;      /* from url or property */
+  gchar            *prop_proxy_id;     /* set via property */
+  gchar            *prop_proxy_pw;     /* set via property */
+  guint             rtp_blocksize;
+  gchar            *user_id;
+  gchar            *user_pw;
+  gint              buffer_mode;
+  GstRTSPRange      client_port_range;
+  gint              udp_buffer_size;
+  gboolean          short_header;
+  guint             probation;
+  gboolean          udp_reconnect;
+  gchar            *multi_iface;
+  gboolean          ntp_sync;
+  gboolean          use_pipeline_clock;
+  GstStructure     *sdes;
+  GTlsCertificateFlags tls_validation_flags;
+  GTlsDatabase     *tls_database;
+  gboolean          do_retransmission;
+
+  /* state */
+  GstRTSPState       state;
+  gchar             *content_base;
+  GstRTSPLowerTrans  cur_protocols;
+  gboolean           tried_url_auth;
+  gchar             *addr;
+  gboolean           need_redirect;
+  GstRTSPTimeRange  *range;
+  gchar             *control;
+  guint              next_port_num;
+  GstClock          *provided_clock;
+
+  /* supported methods */
+  gint               methods;
+
+  gboolean           seekable;
+  GstClockTime       last_pos;
+
+  /* session management */
+  GstElement      *manager;
+  gulong           manager_sig_id;
+  gulong           manager_ptmap_id;
+  gboolean         use_buffering;
+
+  GstRTSPConnInfo  conninfo;
+
+  /* a list of RTSP extensions as GstElement */
+  GstRTSPExtensionList  *extensions;
+
+#ifdef GST_EXT_RTSP_MODIFICATION
+  GCond open_end;
+  GMutex pause_lock;
+  guint64 start_position;
+#endif
+};
+
+struct _GstRTSPSrcClass {
+  GstBinClass parent_class;
+};
+
+GType gst_rtspsrc_get_type(void);
+
+G_END_DECLS
+
+#endif /* __GST_RTSPSRC_H__ */