最近我从他们的站点下载了 Live555 服务器源代码。我尝试编译并运行 testProgs 目录中的 testMPEG1or2AudioVideoStreamer.cpp 文件。我成功地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。运行后得到如下输出:
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 VLC 只是等待一段时间然后停止(Gnome MPlayer 也一样)。它不播放任何音频或视频。感谢您提供任何帮助，因为如果不使用 Live555 成功进行流式传输，我将无法继续。以下是 testMPEG1or2AudioVideoStreamer.cpp 的代码。你能告诉我我错过了什么吗...
/**********
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 是一个 tcp 端口并且是开放的。但它显示应用程序为 freeciv，但我还没有在我的系统上安装该游戏。可能是 Zenmap 的猜测。我在我的系统上运行 Fedora 16 和 gnome 3.2。
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
一切正常，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
最佳答案
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 554，8554 作为辅助 IIRC。
看起来您修改了服务器上的 live555 代码以改为使用 5555。但是你不知道 VLC 对 live555 的使用是否支持使用非标准的 RTSP 端口号。我想您可以在 VLC 代码中查找它。
你可以尝试的事情:
- 使用 openRTSP 与 URI 一起工作
- 使用数据包嗅探器查看网络上实际发生的情况，即正在使用哪些端口。
- 使用标准端口看看是否可行
这些步骤可以让您缩小问题所在的范围。
编辑:
从 RTSP 通信中您可以看到 VLC 正在尝试创建单播 session ，服务器使用多播传输地址进行响应。然后 VLC 播放流，在 10 秒内没有接收到任何数据，然后尝试启动交错的 RTP over RTSP session ，服务器再次以多播地址响应该 session ，因此 RTSP 服务器以 461 响应。 根据 live555:
testMPEG1or2AudioVideoStreamer reads a MPEG-1 or 2 Program Stream file (named "test.mpg"), extracts from this an audio and a video Elementary Stream, and streams these, using RTP, to the multicast group 239.255.42.42, port 6666/6667 (for the audio stream) and 8888/8889 (for the video stream). This program also has an (optional) built-in RTSP server.
关于c++ - 无法使用 Live555 服务器进行流式传输 - 示例不起作用，我们在Stack Overflow上找到一个类似的问题： https://stackoverflow.com/questions/8316697/