[Live-devel] Error in testMPEG4VideoStreamer
Hi Ross, I was testing your testMPEG4VideoStreamer, but it gives error 1272415941.241840 Groupsock(6028: 232.0.4.12, 1, 255): failed to join group: setsockopt(IP_ADD_MEMBERSHIP) error: Unknown error 1272415941.243164 Groupsock(5940: 232.0.4.12, 18889, 255): failed to join group: setsockopt(IP_ADD_MEMBERSHIP) error: Unknown error I googled it and found that i need to set route to 224.0.0/4. I am using windows XP sp 2 and by using route print command, i found that the route has already been added to the table for 224.0.0/4. My pc is on LAN. So how to solve this problem. Thanks Vikas ___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel
[Live-devel] Re :: initialize with URL
Hi Ross, Thanks a ton for your reply. ys,i also thought and have done that but that was not working as a whole.Actually my requirement is to stream live source in H.264 and as you said in FAQ ,i am making a subclass of OnDemandMediaSubsession where i have to redefine virtual function CreateNewStreamSource() and have to return FramedSource * from inside this. For this i have followed openRTSP 1) Created RTSPClient then provide URL with describeURL() method 2) Created MediaSession 3) Iterated through MediaSession and from Video subsession (i have just one SubSession Video) returned rtpSource as mediaSubsessoin-rtpSource() (after initializing) But i was getting error , is this method is correct to do such type of thing inside createNewStreamSource() or i have to do StreamFramer type of thing. Just hint will help me a lot to go in right direction. Thanks Vikas ** I have to define H264VideoRTPSource from URL instead of groupSock , ip/port , is it possible to do this. If the URL is a rtsp:// URL, then just use openRTSP. -- Ross Finlayson Live Networks, Inc. http://www.live555.com/ Your Mail works best with the New Yahoo Optimized IE8. Get it NOW! http://downloads.yahoo.com/in/internetexplorer/___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel
[Live-devel] initialize with URL
Hi , I have to define H264VideoRTPSource from URL instead of groupSock , ip/port , is it possible to do this.If yes then , what should i do , any little hint will help me a lot. Thanks Vikas The INTERNET now has a personality. YOURS! See your Yahoo! Homepage. http://in.yahoo.com/___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel
[Live-devel] H.264 Streaming from Live H.264 source
Hi Developers,
I have requirement to create streaming in H.264 , i am receiving stream from
live source(in H.264 format) and i have to re-stream it to another ip/port.
for that i have edited openRTSP and added one more option for re-streaming and
added following code.
RTPSink* videoSink;
iter.reset();
{while((subsession = iter.next()) != NULL) //***Create our RTPsink
variables...// Create 'groupsocks' for RTP and RTCP:charconst*
destinationAddressStr = 10.69.169.149;constunsignedshortrtpPortNum =
;constunsignedshortrtcpPortNum = rtpPortNum+1;constunsignedcharttl = 7; //
low, in case routers don't admin scopedestinationAddress.s_addr =
our_inet_addr(destinationAddressStr);
Groupsock rtpGroupsock(*env, destinationAddress, rtpPort, ttl);
Groupsock rtcpGroupsock(*env, destinationAddress, rtcpPort, ttl);structin_addr
destinationAddress;constPort rtpPort(rtpPortNum);constPort
rtcpPort(rtcpPortNum);//Create our sink...videoSink =
H264VideoRTPSink::createNew(*env,
rtpGroupsock,subsession-rtpPayloadFormat(),subsession-fmtp_profile_level_id(),subsession-fmtp_spropparametersets());
*env
exit(1);
}if(videoSink == NULL){Unable to create sink
\n;//** subsession-sink =
videoSink;
subsession-sink-startPlaying(*(subsession-readSource()),subsessionAfterPlaying,subsession);
}
But i am unable to play stream , i am using VLC player.
Can anybody help me out,
for rtpPayloadFormat,profile_level_id,sprop_parameter_sets_str in
H264VideoRTPSink::createNew , i am putting whatever i am getting from live
source.
Thanks
Vikas Srivastava
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[Live-devel] Is this correct code
Hi Ross,
I am using your Live555 and facing some difficulties , i am receiving
stream over rtp and saving to file , following is the code
int
TaskScheduler* scheduler = BasicTaskScheduler::createNew();
UsageEnvironment*env = BasicUsageEnvironment::createNew(*scheduler);
main(intargc, char** argv) {
FileSink* fileSink;//Dot specify codecfileSink =FileSink::createNew(*env,
test);
//**For RTP Stream receiving
**sessionAddress.s_addr =
our_inet_addr(sessionAddressStr);
Groupsock rtpGroupsock(*env, sessionAddress, rtpPort, ttl);
Groupsock rtcpGroupsock(*env, sessionAddress, rtcpPort, ttl);
source = MPEG1or2VideoRTPSource::createNew(*env, rtpGroupsock); charconst*
sessionAddressStr= 10.69.169.149;constunsignedshortrtpPortNum =
;constunsignedshortrtcpPortNum = rtpPortNum+1;constunsignedcharttl =
2;structin_addr sessionAddress;constPort rtpPort(rtpPortNum);constPort
rtcpPort(rtcpPortNum);constunsignedestimatedSessionBandwidth = 160; // in kbps;
for RTCP b/w sharegethostname((
CNAME[maxCNAMElen] = constunsignedmaxCNAMElen =
100;unsignedcharCNAME[maxCNAMElen+1];char*)CNAME, maxCNAMElen);'\0'; // just in
caseRTCPInstance* rtcpInstance=RTCPInstance::createNew(*env, rtcpGroupsock,
estimatedSessionBandwidth, CNAME,
NULL
fileSink-startPlaying(*source, afterPlaying, NULL);
env-taskScheduler().doEventLoop();return0;
}
voidafterPlaying(void* /*clientData*/) {//Medium::close(videoSource);//For
RTPMedium::close(source);
}
this code never give any error but still not saving inside file , file size
is always to zero .
Pls help me out wht might be the reason.
Thaanks
Vikas/* we're a client */, source);
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[Live-devel] Saved H.264 file doent Play
Hi Ross, I am receiving stream from server which sends stream in H.264 format over RTSP , i used your openRTSP to save the stream but saved file doent show Video (I used VLC) , but when i received stream in MPeg1 or 2 format then i could see the file with VLC play. What is problem with H.264 format , i saw you are using sperate H264VideoFileSink class.So where might be the problem , with the stream or openRTSP. Thanks Vikas The INTERNET now has a personality. YOURS! See your Yahoo! Homepage. The INTERNET now has a personality. YOURS! See your Yahoo! Homepage. http://in.yahoo.com/___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel
[Live-devel] RTSP to UDP
Hi Ross, I am new to your Live555 library so do you have any sample code from receiving in RTSP and then re streaming over UDP.I got sample code for receiving in RTSP but how again re stream it over UDP or RTSP instead of saving to file , did't get. Thanks Vikas The INTERNET now has a personality. YOURS! See your Yahoo! Homepage. http://in.yahoo.com/___ live-devel mailing list live-devel@lists.live555.com http://lists.live555.com/mailman/listinfo/live-devel
