Hi All, I am trying to write a sip application for audio/video call using
codec PCMU and H.263.I negotiate the SDP parameter with kapanga(sip softphone)
and then call the Live555 API for sending the H.263 video.But somehow on
receiving the H.263 video stream, Kapnga shows "Bad marker" and "header
damaged" on reception of H263 packet. I negotiate with kapanga using
"h263-1998" and payload type 105 .Following is my code snippet for starting the
video call:(this is almost same what has been given in live555's test program)
int PlayVideo(char *ipAddr,int videoRtpPort,int videoRtcpPort) { // 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 //= "10.0.8.32"; =ipAddr;#else =
"239.255.42.42"; // 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 rtpPortNum
=videoRtpPort; // 8888; const unsigned short
rtcpPortNum = videoRtcpPort; const unsigned char ttl = 255; // low, in case
routers don't admin scope
struct in_addr destinationAddress; destinationAddress.s_addr =
our_inet_addr(destinationAddressStr); const Port rtpPort(rtpPortNum); const
Port rtcpPort(rtcpPortNum);
Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl); Groupsock
rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);#ifdef USE_SSM
rtpGroupsock.multicastSendOnly(); rtcpGroupsock.multicastSendOnly();#endif
// Create a 'MPEG Video RTP' sink from the RTP 'groupsock': //videoSink =
MPEG1or2VideoRTPSink::createNew(*env, &rtpGroupsock); videoSink
=H263plusVideoRTPSink::createNew(*env, &rtpGroupsock,105);
// Create (and start) a 'RTCP instance' for this RTP sink: const unsigned
estimatedSessionBandwidth = 500; // 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* rtcp =#endif
RTCPInstance::createNew(*env, &rtcpGroupsock,
estimatedSessionBandwidth, CNAME, videoSink, NULL
/* we're a server */, isSSM); // Note: This starts RTCP running automatically
#ifdef IMPLEMENT_RTSP_SERVER RTSPServer* rtspServer =
RTSPServer::createNew(*env); // 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 \"testMPEG1or2VideoStreamer\"",
isSSM);
sms->addSubsession(PassiveServerMediaSubsession::createNew(*videoSink, rtcp));
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}
I am also attaching the media file which I am using for streaming.Can anyone
please help me for pointing out the issue ?
Best Regards,Parveen Jain
Build a bright career through MSN Education Sign up now.
_________________________________________________________________
Bollywood, beauties and the latest flicks on MSN entertainment
http://entertainment.in.msn.com/
_______________________________________________
live-devel mailing list
live-devel@lists.live555.com
http://lists.live555.com/mailman/listinfo/live-devel