title: Cannot stream with Live555 server - Example not working
body: <p>Recently I downloaded Live555 server source code from their site. I tried to compile and run <code>testMPEG1or2AudioVideoStreamer.cpp</code> file in the <code>testProgs</code> directory. I compiled the whole project including the test programs successfully. Then I run the <code>testMPEG1or2AudioVideoStreamer</code> test program. I also placed a <code>test.mpg</code> file in the current directory as defined in the test program. After running I got the following output:</p>
  
  <pre><code>Play this stream using the URL "rtsp://192.168.2.22:5555/testStream"
  Beginning streaming...
  Beginning to read from file...
  ...done reading from file
  Beginning to read from file...
  ...done reading from file
  etc.,
  </code></pre>
  
  <p>Then I copy and play the URL <code>rtsp://192.168.2.22:5555/testStream</code> using VLC media player, but VLC just wait sometime and then stop (same with Gnome MPlayer also). It does not play any audio or video. Any help is appreciated as I cannot go forward without successfully streaming using Live555. Here is the code of <code>testMPEG1or2AudioVideoStreamer.cpp</code>. Can you tell me what am I missing...</p>
  
  <pre><code>/**********
  This library is free software; you can redistribute it and/or modify it under
  the terms of the GNU Lesser General Public License as published by the
  Free Software Foundation; either version 2.1 of the License, or (at your
  option) any later version. (See &lt;http://www.gnu.org/copyleft/lesser.html&gt;.)
  
  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 Lesser General Public License for
  more details.
  
  You should have received a copy of the GNU Lesser General Public License
  along with this library; if not, write to the Free Software Foundation, Inc.,
  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301  USA
  **********/
  // Copyright (c) 1996-2010, Live Networks, Inc.  All rights reserved
  // A test program that reads a MPEG-1 or 2 Program Stream file,
  // splits it into Audio and Video Elementary Streams,
  // and streams both using RTP
  // main program
  
  #include "liveMedia.hh"
  #include "BasicUsageEnvironment.hh"
  #include "GroupsockHelper.hh"
  
  UsageEnvironment* env;
  char const* inputFileName = "test.mpg";
  MPEG1or2Demux* mpegDemux;
  FramedSource* audioSource;
  FramedSource* videoSource;
  RTPSink* audioSink;
  RTPSink* videoSink;
  
  void play(); // forward
  
  // To stream using "source-specific multicast" (SSM), uncomment the following:
  //#define USE_SSM 1
  #ifdef USE_SSM
  Boolean const isSSM = True;
  #else
  Boolean const isSSM = False;
  #endif
  
  // To set up an internal RTSP server, uncomment the following:
  #define IMPLEMENT_RTSP_SERVER 1
  // (Note that this RTSP server works for multicast only)
  
  // To stream *only* MPEG "I" frames (e.g., to reduce network bandwidth),
  // change the following "False" to "True":
  Boolean iFramesOnly = False;
  
  int main(int argc, char** argv) {
    // Begin by setting up our usage environment:
    TaskScheduler* scheduler = BasicTaskScheduler::createNew();
    env = BasicUsageEnvironment::createNew(*scheduler);
  
    // Create 'groupsocks' for RTP and RTCP:
    char const* destinationAddressStr
  #ifdef USE_SSM
      = "192.168.1.255";
  #else
      = "192.168.1.255";
    // Note: This is a multicast address.  If you wish to stream using
    // unicast instead, then replace this string with the unicast address
    // of the (single) destination.  (You may also need to make a similar
    // change to the receiver program.)
  #endif
    const unsigned short rtpPortNumAudio = 6666;
    const unsigned short rtcpPortNumAudio = rtpPortNumAudio+1;
    const unsigned short rtpPortNumVideo = 8888;
    const unsigned short rtcpPortNumVideo = rtpPortNumVideo+1;
    const unsigned char ttl = 7; // low, in case routers don't admin scope
  
    struct in_addr destinationAddress;
    destinationAddress.s_addr = our_inet_addr(destinationAddressStr);
    const Port rtpPortAudio(rtpPortNumAudio);
    const Port rtcpPortAudio(rtcpPortNumAudio);
    const Port rtpPortVideo(rtpPortNumVideo);
    const Port rtcpPortVideo(rtcpPortNumVideo);
  
    Groupsock rtpGroupsockAudio(*env, destinationAddress, rtpPortAudio, ttl);
    Groupsock rtcpGroupsockAudio(*env, destinationAddress, rtcpPortAudio, ttl);
    Groupsock rtpGroupsockVideo(*env, destinationAddress, rtpPortVideo, ttl);
    Groupsock rtcpGroupsockVideo(*env, destinationAddress, rtcpPortVideo, ttl);
  #ifdef USE_SSM
    rtpGroupsockAudio.multicastSendOnly();
    rtcpGroupsockAudio.multicastSendOnly();
    rtpGroupsockVideo.multicastSendOnly();
    rtcpGroupsockVideo.multicastSendOnly();
  #endif
  
    // Create a 'MPEG Audio RTP' sink from the RTP 'groupsock':
    audioSink = MPEG1or2AudioRTPSink::createNew(*env, &amp;rtpGroupsockAudio);
  
    // Create (and start) a 'RTCP instance' for this RTP sink:
    const unsigned estimatedSessionBandwidthAudio = 160; // in kbps; for RTCP b/w share
    const unsigned maxCNAMElen = 100;
    unsigned char CNAME[maxCNAMElen+1];
    gethostname((char*)CNAME, maxCNAMElen);
    CNAME[maxCNAMElen] = '\0'; // just in case
  #ifdef IMPLEMENT_RTSP_SERVER
    RTCPInstance* audioRTCP =
  #endif
      RTCPInstance::createNew(*env, &amp;rtcpGroupsockAudio,
                  estimatedSessionBandwidthAudio, CNAME,
                  audioSink, NULL /* we're a server */, isSSM);
    // Note: This starts RTCP running automatically
  
    // Create a 'MPEG Video RTP' sink from the RTP 'groupsock':
    videoSink = MPEG1or2VideoRTPSink::createNew(*env, &amp;rtpGroupsockVideo);
  
    // Create (and start) a 'RTCP instance' for this RTP sink:
    const unsigned estimatedSessionBandwidthVideo = 4500; // in kbps; for RTCP b/w share
  #ifdef IMPLEMENT_RTSP_SERVER
    RTCPInstance* videoRTCP =
  #endif
      RTCPInstance::createNew(*env, &amp;rtcpGroupsockVideo,
                    estimatedSessionBandwidthVideo, CNAME,
                    videoSink, NULL /* we're a server */, isSSM);
    // Note: This starts RTCP running automatically
  
  #ifdef IMPLEMENT_RTSP_SERVER
    RTSPServer* rtspServer = RTSPServer::createNew(*env, 5555);
    // Note that this (attempts to) start a server on the default RTSP server
    // port: 554.  To use a different port number, add it as an extra
    // (optional) parameter to the "RTSPServer::createNew()" call above.
    if (rtspServer == NULL) {
      *env &lt;&lt; "Failed to create RTSP server: " &lt;&lt; env-&gt;getResultMsg() &lt;&lt; "\n";
      exit(1);
    }
    ServerMediaSession* sms
      = ServerMediaSession::createNew(*env, "testStream", inputFileName,
             "Session streamed by \"testMPEG1or2AudioVideoStreamer\"",
                         isSSM);
    sms-&gt;addSubsession(PassiveServerMediaSubsession::createNew(*audioSink, audioRTCP));
    sms-&gt;addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, videoRTCP));
    rtspServer-&gt;addServerMediaSession(sms);
  
    char* url = rtspServer-&gt;rtspURL(sms);
    *env &lt;&lt; "Play this stream using the URL \"" &lt;&lt; url &lt;&lt; "\"\n";
    delete[] url;
  #endif
  
    // Finally, start the streaming:
    *env &lt;&lt; "Beginning streaming...\n";
    play();
  
    env-&gt;taskScheduler().doEventLoop(); // does not return
  
    return 0; // only to prevent compiler warning
  }
  
  void afterPlaying(void* clientData) {
    // One of the sinks has ended playing.
    // Check whether any of the sources have a pending read.  If so,
    // wait until its sink ends playing also:
    if (audioSource-&gt;isCurrentlyAwaitingData()
        || videoSource-&gt;isCurrentlyAwaitingData()) return;
  
    // Now that both sinks have ended, close both input sources,
    // and start playing again:
    *env &lt;&lt; "...done reading from file\n";
  
    audioSink-&gt;stopPlaying();
    videoSink-&gt;stopPlaying();
        // ensures that both are shut down
    Medium::close(audioSource);
    Medium::close(videoSource);
    Medium::close(mpegDemux);
    // Note: This also closes the input file that this source read from.
  
    // Start playing once again:
    play();
  }
  
  void play() {
    // Open the input file as a 'byte-stream file source':
    ByteStreamFileSource* fileSource
      = ByteStreamFileSource::createNew(*env, inputFileName);
    if (fileSource == NULL) {
      *env &lt;&lt; "Unable to open file \"" &lt;&lt; inputFileName
       &lt;&lt; "\" as a byte-stream file source\n";
      exit(1);
    }
  
    // We must demultiplex Audio and Video Elementary Streams
    // from the input source:
    mpegDemux = MPEG1or2Demux::createNew(*env, fileSource);
    FramedSource* audioES = mpegDemux-&gt;newAudioStream();
    FramedSource* videoES = mpegDemux-&gt;newVideoStream();
  
    // Create a framer for each Elementary Stream:
    audioSource
      = MPEG1or2AudioStreamFramer::createNew(*env, audioES);
    videoSource
      = MPEG1or2VideoStreamFramer::createNew(*env, videoES, iFramesOnly);
  
    // Finally, start playing each sink.
    *env &lt;&lt; "Beginning to read from file...\n";
    videoSink-&gt;startPlaying(*videoSource, afterPlaying, videoSink);
    audioSink-&gt;startPlaying(*audioSource, afterPlaying, audioSink);
  }
  </code></pre>
  
  <p><strong>EDIT 1:</strong> <code>openRTSP</code> output</p>
  
  <pre><code>[jomit@jomoos live2]$ testProgs/openRTSP -o rtsp://192.168.2.22:5555/testStream
  Sending request: OPTIONS rtsp://192.168.2.22:5555/testStream RTSP/1.0
  CSeq: 1
  User-Agent: testProgs/openRTSP (LIVE555 Streaming Media v2010.03.08)
  
  
  Received OPTIONS response: RTSP/1.0 200 OK
  CSeq: 1
  Date: Wed, Nov 30 2011 08:30:23 GMT
  Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, SET_PARAMETER
  
  
  RTSP "OPTIONS" request returned: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE,     SET_PARAMETER
  </code></pre>
  
  <p><strong>EDIT 2:</strong> port check</p>
  
  <p>I used Zenmap to scan the ports, and it shows 5555 as a tcp port and as open. But it shows the application as freeciv, but I haven't installed that game on my system. May be it is a guess by Zenmap. I am running Fedora 16 with gnome 3.2 on my system.</p>
  
  <p><strong>EDIT 3:</strong> VLC output</p>
  
  <pre><code>[0x21fa840] main playlist debug: processing request item rtsp://192.168.1.222:5555/testStream node Playlist skip 0
  [0x21fa840] main playlist debug: resyncing on rtsp://192.168.1.222:5555/testStream
  [0x21fa840] main playlist debug: rtsp://192.168.1.222:5555/testStream is at 0
  [0x21fa840] main playlist debug: starting new item
  [0x21fa840] main playlist debug: creating new input thread
  [0x7f1f88005410] main input debug: Creating an input for 'rtsp://192.168.1.222:5555/testStream'
  [0x7f1f88005410] main input debug: thread (input) created at priority 10 (input/input.c:220)
  [0x7f1f88005ec0] main input debug: TIMER input launching for 'rtsp://192.168.1.222:5555/testStream' : 15.307 ms - Total 15.307 ms / 1 intvls (Avg 15.307 ms)
  [0x2227990] qt4 interface debug: IM: Setting an input
  [0x7f1f88005410] main input debug: thread started
  [0x7f1f88005410] main input debug: using timeshift granularity of 50 MiB
  [0x7f1f88005410] main input debug: using timeshift path '/tmp'
  [0x7f1f88005410] main input debug: `rtsp://192.168.1.222:5555/testStream' gives access `rtsp' demux `' path `192.168.1.222:5555/testStream'
  [0x7f1f88005410] main input debug: creating demux: access='rtsp' demux='' path='192.168.1.222:5555/testStream'
  [0x7f1f7c002860] main demux debug: looking for access_demux module: 1 candidate
  Opening connection to 192.168.1.222, port 5555...
  ...remote connection opened
  Sending request: OPTIONS rtsp://192.168.1.222:5555/testStream RTSP/1.0
  CSeq: 2
  User-Agent: LibVLC/1.1.12 (LIVE555 Streaming Media v2011.09.02)
  
  
  Received 137 new bytes of response data.
  Received a complete OPTIONS response:
  RTSP/1.0 200 OK
  CSeq: 2
  Date: Wed, Nov 30 2011 19:45:55 GMT
  Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, SET_PARAMETER
  
  
  Sending request: DESCRIBE rtsp://192.168.1.222:5555/testStream RTSP/1.0
  CSeq: 3
  User-Agent: LibVLC/1.1.12 (LIVE555 Streaming Media v2011.09.02)
  Accept: application/sdp
  
  
  Received 641 new bytes of response data.
  Received a complete DESCRIBE response:
  RTSP/1.0 200 OK
  CSeq: 3
  Date: Wed, Nov 30 2011 19:45:55 GMT
  Content-Base: rtsp://192.168.1.222:5555/testStream/
  Content-Type: application/sdp
  Content-Length: 471
  
  v=0
  o=- 1322681211098021 1 IN IP4 192.168.1.222
  s=Session streamed by "testMPEG1or2AudioVideoStreamer"
  i=test.mpg
  t=0 0
  a=tool:LIVE555 Streaming Media v2010.03.08
  a=type:broadcast
  a=control:*
  a=range:npt=0-
  a=x-qt-text-nam:Session streamed by "testMPEG1or2AudioVideoStreamer"
  a=x-qt-text-inf:test.mpg
  m=audio 6666 RTP/AVP 14
  c=IN IP4 192.168.1.255/7
  b=AS:160
  a=control:track1
  m=video 8888 RTP/AVP 32
  c=IN IP4 192.168.1.255/7
  b=AS:4500
  a=control:track2
  
  [0x7f1f7c002860] live555 demux debug: RTP subsession 'audio/MPA'
  Sending request: SETUP rtsp://192.168.1.222:5555/testStream/track1 RTSP/1.0
  CSeq: 4
  User-Agent: LibVLC/1.1.12 (LIVE555 Streaming Media v2011.09.02)
  Transport: RTP/AVP;unicast;client_port=6666-6667
  
  
  Received 182 new bytes of response data.
  Received a complete SETUP response:
  RTSP/1.0 200 OK
  CSeq: 4
  Date: Wed, Nov 30 2011 19:45:55 GMT
  Transport: RTP/AVP;multicast;destination=192.168.1.255;source=192.168.1.222;port=6666-6667;ttl=7
  Session: 06AFB6E5
  
  
  [0x7f1f88005410] main input debug: selecting program id=0
  [0x7f1f7c002860] live555 demux debug: RTP subsession 'video/MPV'
  Sending request: SETUP rtsp://192.168.1.222:5555/testStream/track2 RTSP/1.0
  CSeq: 5
  User-Agent: LibVLC/1.1.12 (LIVE555 Streaming Media v2011.09.02)
  Transport: RTP/AVP;unicast;client_port=8888-8889
  Session: 06AFB6E5
  
  
  Received 182 new bytes of response data.
  Received a complete SETUP response:
  RTSP/1.0 200 OK
  CSeq: 5
  Date: Wed, Nov 30 2011 19:45:55 GMT
  Transport: RTP/AVP;multicast;destination=192.168.1.255;source=192.168.1.222;port=8888-8889;ttl=7
  Session: 06AFB6E5
  
  
  [0x7f1f7c002860] live555 demux debug: setup start: 0.000000 stop:0.000000
  Sending request: PLAY rtsp://192.168.1.222:5555/testStream/ RTSP/1.0
  CSeq: 6
  User-Agent: LibVLC/1.1.12 (LIVE555 Streaming Media v2011.09.02)
  Session: 06AFB6E5
  Range: npt=0.000-
  
  
  Received 268 new bytes of response data.
  Received a complete PLAY response:
  RTSP/1.0 200 OK
  CSeq: 6
  Date: Wed, Nov 30 2011 19:45:55 GMT
  Range: npt=0.000-
  Session: 06AFB6E5
  RTP-Info: url=rtsp://192.168.1.222:5555/testStream/track1;seq=33348;rtptime=3573241747,url=rtsp://192.168.1.222:5555/testStream/track2;seq=12520;rtptime=2773558772
  
  
  [0x7f1f7c002860] live555 demux debug: play start: 0.000000 stop:0.000000
  [0x7f1f7c002860] main demux debug: using access_demux module "live555"
  [0x7f1f7c002860] main demux debug: TIMER module_need() : 5.536 ms - Total 5.536 ms / 1 intvls (Avg 5.536 ms)
  [0x7f1f7c00dca0] main decoder debug: looking for decoder module: 33 candidates
  [0x7f1f7c00dca0] main decoder debug: using decoder module "mpeg_audio"
  [0x7f1f7c00dca0] main decoder debug: TIMER module_need() : 0.519 ms - Total 0.519 ms / 1 intvls (Avg 0.519 ms)
  [0x7f1f7c00dca0] main decoder debug: thread (decoder) created at priority 5 (input/decoder.c:301)
  [0x7f1f7c00dca0] main decoder debug: thread started
  [0x7f1f7c00e5f0] main decoder debug: looking for decoder module: 33 candidates
  [0x7f1f7c00e5f0] avcodec decoder debug: libavcodec already initialized
  [0x7f1f7c00e5f0] avcodec decoder debug: trying to use direct rendering
  [0x7f1f7c00e5f0] avcodec decoder debug: ffmpeg codec (MPEG-1/2 Video) started
  [0x7f1f7c00e5f0] main decoder debug: using decoder module "avcodec"
  [0x7f1f7c00e5f0] main decoder debug: TIMER module_need() : 1.561 ms - Total 1.561 ms / 1 intvls (Avg 1.561 ms)
  [0x7f1f7c006b90] main packetizer debug: looking for packetizer module: 21 candidates
  [0x7f1f7c006b90] main packetizer debug: using packetizer module "packetizer_mpegvideo"
  [0x7f1f7c006b90] main packetizer debug: TIMER module_need() : 0.288 ms - Total 0.288 ms / 1 intvls (Avg 0.288 ms)
  [0x7f1f7c00e5f0] main decoder debug: thread (decoder) created at priority 0 (input/decoder.c:301)
  [0x7f1f7c00e5f0] main decoder debug: thread started
  [0x7f1f7c008250] main demux meta debug: looking for meta reader module: 2 candidates
  [0x7f1f7c008250] lua demux meta debug: Trying Lua scripts in /home/jomit/.local/share/vlc/lua/meta/reader
  [0x7f1f7c008250] lua demux meta debug: Trying Lua scripts in /usr/lib64/vlc/lua/meta/reader
  [0x7f1f7c008250] lua demux meta debug: Trying Lua playlist script /usr/lib64/vlc/lua/meta/reader/filename.luac
  [0x7f1f7c008250] lua demux meta debug: Trying Lua scripts in /usr/share/vlc/lua/meta/reader
  [0x7f1f7c008250] main demux meta debug: no meta reader module matching "any" could be loaded
  [0x7f1f7c008250] main demux meta debug: TIMER module_need() : 1.093 ms - Total 1.093 ms / 1 intvls (Avg 1.093 ms)
  [0x7f1f88005410] main input debug: `rtsp://192.168.1.222:5555/testStream' successfully opened
  [0x7f1f7c002860] live555 demux warning: no data received in 10s. Switching to TCP
  Sending request: TEARDOWN rtsp://192.168.1.222:5555/testStream/ RTSP/1.0
  CSeq: 7
  User-Agent: LibVLC/1.1.12 (LIVE555 Streaming Media v2011.09.02)
  Session: 06AFB6E5
  
  
  [0x7f1f7c00dca0] main decoder debug: removing module "mpeg_audio"
  [0x7f1f7c00dca0] main decoder debug: killing decoder fourcc `mpga', 0 PES in FIFO
  [0x7f1f7c00e5f0] avcodec decoder debug: ffmpeg codec (MPEG-1/2 Video) stopped
  [0x7f1f7c00e5f0] main decoder debug: removing module "avcodec"
  [0x7f1f7c00e5f0] main decoder debug: killing decoder fourcc `mpgv', 0 PES in FIFO
  [0x7f1f7c006b90] main packetizer debug: removing module "packetizer_mpegvideo"
  [0x7f1f88005410] main input debug: Program doesn't contain anymore ES
  Opening connection to 192.168.1.222, port 5555...
  ...remote connection opened
  Sending request: OPTIONS rtsp://192.168.1.222:5555/testStream RTSP/1.0
  CSeq: 2
  User-Agent: LibVLC/1.1.12 (LIVE555 Streaming Media v2011.09.02)
  
  
  Received 137 new bytes of response data.
  Received a complete OPTIONS response:
  RTSP/1.0 200 OK
  CSeq: 2
  Date: Wed, Nov 30 2011 19:46:05 GMT
  Public: OPTIONS, DESCRIBE, SETUP, TEARDOWN, PLAY, PAUSE, SET_PARAMETER
  
  
  Sending request: DESCRIBE rtsp://192.168.1.222:5555/testStream RTSP/1.0
  CSeq: 3
  User-Agent: LibVLC/1.1.12 (LIVE555 Streaming Media v2011.09.02)
  Accept: application/sdp
  
  
  Received 641 new bytes of response data.
  Received a complete DESCRIBE response:
  RTSP/1.0 200 OK
  CSeq: 3
  Date: Wed, Nov 30 2011 19:46:05 GMT
  Content-Base: rtsp://192.168.1.222:5555/testStream/
  Content-Type: application/sdp
  Content-Length: 471
  
  v=0
  o=- 1322681211098021 1 IN IP4 192.168.1.222
  s=Session streamed by "testMPEG1or2AudioVideoStreamer"
  i=test.mpg
  t=0 0
  a=tool:LIVE555 Streaming Media v2010.03.08
  a=type:broadcast
  a=control:*
  a=range:npt=0-
  a=x-qt-text-nam:Session streamed by "testMPEG1or2AudioVideoStreamer"
  a=x-qt-text-inf:test.mpg
  m=audio 6666 RTP/AVP 14
  c=IN IP4 192.168.1.255/7
  b=AS:160
  a=control:track1
  m=video 8888 RTP/AVP 32
  c=IN IP4 192.168.1.255/7
  b=AS:4500
  a=control:track2
  
  [0x7f1f7c002860] live555 demux debug: RTP subsession 'audio/MPA'
  Sending request: SETUP rtsp://192.168.1.222:5555/testStream/track1 RTSP/1.0
  CSeq: 4
  User-Agent: LibVLC/1.1.12 (LIVE555 Streaming Media v2011.09.02)
  Transport: RTP/AVP/TCP;unicast;interleaved=0-1
  
  
  Received 84 new bytes of response data.
  Received a complete SETUP response:
  RTSP/1.0 461 Unsupported Transport
  CSeq: 4
  Date: Wed, Nov 30 2011 19:46:05 GMT
  
  
  Sending request: SETUP rtsp://192.168.1.222:5555/testStream/track1 RTSP/1.0
  CSeq: 5
  User-Agent: LibVLC/1.1.12 (LIVE555 Streaming Media v2011.09.02)
  Transport: RTP/AVP;unicast;client_port=6666-6667
  
  
  [0x7f1f7c002860] live555 demux error: SETUP of'audio/MPA' failed 461 Unsupported Transport
  [0x7f1f7c002860] live555 demux debug: RTP subsession 'video/MPV'
  Opening connection to 192.168.1.222, port 5555...
  ...remote connection opened
  Sending request: SETUP rtsp://192.168.1.222:5555/testStream/track2 RTSP/1.0
  CSeq: 6
  User-Agent: LibVLC/1.1.12 (LIVE555 Streaming Media v2011.09.02)
  Transport: RTP/AVP/TCP;unicast;interleaved=2-3
  
  
  Received 84 new bytes of response data.
  Received a complete SETUP response:
  RTSP/1.0 461 Unsupported Transport
  CSeq: 6
  Date: Wed, Nov 30 2011 19:46:05 GMT
  
  
  Sending request: SETUP rtsp://192.168.1.222:5555/testStream/track2 RTSP/1.0
  CSeq: 7
  User-Agent: LibVLC/1.1.12 (LIVE555 Streaming Media v2011.09.02)
  Transport: RTP/AVP;unicast;client_port=8888-8889
  
  
  [0x7f1f7c002860] live555 demux error: SETUP of'video/MPV' failed RTSP response was truncated. Increase "RTSPClient::responseBufferSize"
  [0x7f1f7c002860] live555 demux debug: setup start: 0.000000 stop:0.000000
  [0x7f1f7c002860] live555 demux error: Nothing to play for rtsp://192.168.1.222:5555/testStream
  [0x7f1f7c002860] live555 demux error: TCP rollover failed, aborting
  [0x7f1f88005410] main input debug: EOF reached
  [0x21fa840] main playlist debug: finished input
  Opening connection to 192.168.1.222, port 5555...
  [0x7f1f7c002860] main demux debug: removing module "live555"
  [0x7f1f88005410] main input debug: thread ended
  [0x21fa840] main playlist debug: dead input
  [0x21fa840] main playlist debug: changing item without a request (current 0/1)
  [0x21fa840] main playlist debug: nothing to play
  [0x2227990] qt4 interface debug: IM: Deleting the input
  </code></pre>
  
  <p>Everything seems OK, except with the following two errors:</p>
  
  <pre><code>[0x7f1f7c002860] live555 demux error: SETUP of'audio/MPA' failed 461 Unsupported Transport
  </code></pre>
  
  <p>and</p>
  
  <pre><code>[0x7f1f7c002860] live555 demux error: SETUP of'video/MPV' failed RTSP response was truncated. Increase "RTSPClient::responseBufferSize"
  [0x7f1f7c002860] live555 demux debug: setup start: 0.000000 stop:0.000000
  [0x7f1f7c002860] live555 demux error: Nothing to play for rtsp://192.168.1.222:5555/testStream
  [0x7f1f7c002860] live555 demux error: TCP rollover failed, aborting
  </code></pre>