Question

最近我从他们的网站下载了Live555服务器源代码。我试图在testMPEG1or2AudioVideoStreamer.cpp目录中编译并运行testProgs文件。我成功编译了包括测试程序在内的整个项目。然后我运行testMPEG1or2AudioVideoStreamer测试程序。我还在测试程序中定义的当前目录中放置了一个test.mpg文件。运行后，我得到以下输出：

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.,

然后我使用VLC媒体播放器复制并播放URL rtsp://192.168.2.22:5555/testStream，但是VLC只是等待一段时间然后停止（与Gnome MPlayer一样）。它不播放任何音频或视频。任何帮助都表示赞赏，因为如果没有成功使用Live555进行流式传输，我无法前进。这是testMPEG1or2AudioVideoStreamer.cpp的代码。你能告诉我我错过了什么......

/**********
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 (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, &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);
}

编辑1： openRTSP输出

[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

编辑2：端口检查

我使用Zenmap扫描端口，它显示5555作为tcp端口和打开。但它显示应用程序为freeciv，但我没有在我的系统上安装该游戏。可能是Zenmap的猜测。我在我的系统上使用gnome 3.2运行Fedora 16。

编辑3： VLC输出

[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

一切似乎都没问题，除了以下两个错误：

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

和

[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

Answer 1

我怀疑这可能与使用非标准端口号有关，但我可能错了。 IANA分配的RTSP端口为554，而8554为次要IIRC。

看起来你修改了服务器上的live555代码而不是使用5555。但是，您不知道VLC对live555的使用是否支持使用非标准RTSP端口号。我想你可以在VLC代码中看一下。

你可以尝试的事情：

使用openRTSP使用URI
使用数据包嗅探器查看网络上实际发生的情况，即正在使用的端口。
使用标准端口并查看是否有效

这些步骤可让您缩小问题所在。

编辑：

从RTSP通讯中，您可以看到VLC正在尝试创建单播会话，服务器使用多播传输地址进行响应。然后，VLC播放流，不接收10秒的数据，然后尝试在RTSP会话上启动交叉RTP，服务器再次以多播地址响应，因此RTSP服务器以461响应。根据live555：

testMPEG1or2AudioVideoStreamer读取MPEG-1或2节目流文件（名为“test.mpg”），从中提取音频和视频基本流，并使用RTP将这些流传输到组播组239.255.42.42，端口6666/6667（用于音频流）和8888/8889（用于视频流）。该程序还有一个（可选的）内置RTSP服务器。

Answer 2

您有多个网络接口吗？流量可能通过错误的界面。您可以使用Wireshark或其他数据包嗅探器来检查。如果是这种情况，这个邮件线程可能会有所帮助： http://lists.live555.com/pipermail/live-devel/2007-May/006864.html

Answer 3

在我的情况下，禁用虚拟机（在这种情况下为虚拟机）网络适配器有效。

无法使用Live555服务器进行流式处理 - 示例无效

3 个答案: