Không thể truyền trực tuyến với máy chủ Live555 - Ví dụ không hoạt động

Question

Gần đây tôi đã tải xuống mã nguồn máy chủ Live555 từ trang web của họ. Tôi đã cố gắng biên dịch và chạy tệp testMPEG1or2AudioVideoStreamer.cpp trong thư mục testProgs. Tôi đã biên soạn toàn bộ dự án bao gồm các chương trình thử nghiệm thành công. Sau đó, tôi chạy chương trình thử nghiệm testMPEG1or2AudioVideoStreamer. Tôi cũng đặt một tệp test.mpg trong thư mục hiện tại như được xác định trong chương trình thử nghiệm. Sau khi chạy tôi có kết quả như sau:

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., 

Sau đó, tôi sao chép và chơi URL rtsp://192.168.2.22:5555/testStream sử dụng VLC media player, nhưng VLC chỉ đợi lúc nào và sau đó dừng (cùng với Gnome MPlayer cũng). Nó không phát bất kỳ âm thanh hoặc video nào. Bất kỳ trợ giúp nào được đánh giá cao vì tôi không thể tiến lên mà không phát trực tiếp thành công bằng Live555. Đây là mã số testMPEG1or2AudioVideoStreamer.cpp. bạn có thể cho tôi biết những gì tôi thiếu ...

/********** 
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); 
} 

EDIT 1:openRTSP đầu ra

[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: cổng kiểm tra

tôi đã sử dụng Zenmap để quét các cổng và nó hiển thị 5555 như một cổng tcp và mở. Nhưng nó cho thấy các ứng dụng như freeciv, nhưng tôi đã không được cài đặt trò chơi đó trên hệ thống của tôi. Có thể nó là một dự đoán của Zenmap. Tôi đang chạy Fedora 16 với gnome 3.2 trên hệ thống của tôi.

EDIT 3: VLC đầu ra

[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 

Tất cả mọi thứ có vẻ OK, trừ trường hợp được hai lỗi sau:

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

và

[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

Tôi nghi ngờ điều này có thể có một cái gì đó liên quan đến việc sử dụng số cổng không chuẩn, nhưng tôi có thể rong. Cổng RTSP được IANA gán là 554, và 8554 là một IIRC phụ.

Dường như bạn đã sửa đổi mã live555 trên máy chủ để sử dụng 5555 thay thế. Tuy nhiên bạn không biết liệu việc sử dụng live555 của VLC có hỗ trợ sử dụng các số cổng RTSP không chuẩn hay không. Tôi cho rằng bạn có thể xem xét điều này trong mã VLC.

Những điều bạn có thể thử:

• sử dụng openRTSP làm việc với các URI
• sử dụng một gói sniffer để xem những gì đang thực sự xảy ra trên mạng ví dụ: những cổng đang được sử dụng.
• sử dụng cổng tiêu chuẩn và xem nếu mà làm việc

Các bước này sẽ cho phép bạn thu hẹp nơi mà vấn đề là.

Edit:

Từ comms RTSP bạn có thể thấy rằng VLC đang cố gắng để tạo ra một phiên unicast, server trả lời với một địa chỉ vận chuyển multicast.VLC sau đó phát luồng, không nhận dữ liệu trong 10 giây và sau đó cố gắng bắt đầu RTP xen kẽ qua phiên RTSP mà máy chủ lại phản hồi với địa chỉ multicast và do đó máy chủ RTSP phản hồi với 461. Theo live555:

testMPEG1or2AudioVideoStreamer đọc tệp MPEG-1 hoặc 2 Luồng chương trình (có tên là "test.mpg"), trích xuất từ ​​âm thanh này và một luồng video tiểu học và phát các luồng này, sử dụng RTP, tới nhóm multicast 239.255.42.42, cổng 6666/6667 (đối với luồng âm thanh) và 8888/8889 (đối với luồng video). Chương trình này cũng có một máy chủ RTSP tích hợp (tùy chọn).

Answer 2

Bạn có nhiều giao diện mạng không? Lưu lượng truy cập có thể đang đi qua giao diện sai. Bạn có thể sử dụng Wireshark hoặc gói sniffer khác để kiểm tra điều đó. Nếu đúng như vậy, chuỗi thư này có thể hữu ích: http://lists.live555.com/pipermail/live-devel/2007-May/006864.html

Answer 3

Trong trường hợp vô hiệu hóa bộ điều hợp mạng máy ảo (virtualbox trong trường hợp này) hoạt động.