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Unable to stream from Live555 server. The example does not work.

I recently downloaded the source code of the Live555 server from my site. I tried to compile and run the testMPEG1or2AudioVideoStreamer.cpp file in the testMPEG1or2AudioVideoStreamer.cpp directory. I compiled the entire project, including test programs. Then I run the test program testMPEG1or2AudioVideoStreamer . I also placed the test.mpg file in the current directory, as defined in the test program. After starting, I got the following output:

 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.,

Then I copy and play the rtsp://192.168.2.22:5555/testStream using the VLC media player, but the VLC just wait and then stop (same with Gnome MPlayer). It does not play audio or video. Any help is appreciated since I cannot go forward without successful streaming using Live555. Here is the code testMPEG1or2AudioVideoStreamer.cpp . Can you tell me what I am missing ...

 /********** 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 <http://www.gnu.org/copyleft/lesser.html>.) 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 (eg, 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, &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, &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, &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, &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 << "Failed to create RTSP server: " << env->getResultMsg() << "\n"; exit(1); } ServerMediaSession* sms = ServerMediaSession::createNew(*env, "testStream", inputFileName, "Session streamed by \"testMPEG1or2AudioVideoStreamer\"", isSSM); sms->addSubsession(PassiveServerMediaSubsession::createNew(*audioSink, audioRTCP)); sms->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, videoRTCP)); rtspServer->addServerMediaSession(sms); char* url = rtspServer->rtspURL(sms); *env << "Play this stream using the URL \"" << url << "\"\n"; delete[] url; #endif // Finally, start the streaming: *env << "Beginning streaming...\n"; play(); env->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->isCurrentlyAwaitingData() || videoSource->isCurrentlyAwaitingData()) return; // Now that both sinks have ended, close both input sources, // and start playing again: *env << "...done reading from file\n"; audioSink->stopPlaying(); videoSink->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 << "Unable to open file \"" << inputFileName << "\" 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->newAudioStream(); FramedSource* videoES = mpegDemux->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 << "Beginning to read from file...\n"; videoSink->startPlaying(*videoSource, afterPlaying, videoSink); audioSink->startPlaying(*audioSource, afterPlaying, audioSink); }

EDIT 1: openRTSP output

 [ 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

EDIT 2: port check

I used Zenmap to scan ports, and it shows 5555 as tcp port and as open. But it shows the application as freeciv, but I have not installed this game on my system. Maybe this is a guess from Zenmap. I am running Fedora 16 with gnome 3.2 on my system.

EDIT 3: VLC output

 [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

Everything seems OK, except for the following two errors:

 [0x7f1f7c002860] live555 demux error: SETUP of'audio/MPA' failed 461 Unsupported Transport

and

 [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
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3 answers

I suspect this may have something to do with using a non-standard port number, but I could be wrong. The IANA-designated RTSP port is 554, and 8554 is a secondary IIRC.

It looks like you modified the live555 code on the server to use 5555 instead. However, you do not know if VLC use live555 supports the use of non-standard RTSP port numbers. I suppose you could see this in VLC code.

What you can try:

  • use openRTSP to work with URI
  • use the packet sniffer to find out what is actually happening on the network, i.e. which ports are used.
  • use the standard port and see if it works

These steps will allow you to narrow down the problem area.

Edit:

From the RTSP messages, you can see that the VLC is trying to create a unicast session, the server is responding with a multicast transport address. Then the VLC plays the stream, receives no data in 10 seconds, and then tries to start alternating RTP through the RTSP session, to which the server again answers with the multicast address, and therefore the RTSP server answers 461. According to live555:

testMPEG1or2AudioVideoStreamer reads an MPEG-1 or 2 stream file (called "test.mpg"), extracts audio and video from it, and transfers them using RTP to the multicast group 239.255.42.42, port 6666/6667 (for audio stream) and 8888/8889 (for video stream). This program also has an (optional) built-in RTSP server.

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Do you have multiple network interfaces? Traffic may go through the wrong interface. You can use Wireshark or another packet sniffer to test this. In this case, this mail flow may be useful: http://lists.live555.com/pipermail/live-devel/2007-May/006864.html

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In my case, turning off the virtual machine (in this case, the virtual server) worked with network adapters.

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Source: https://habr.com/ru/post/902686/

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