Re: [Live-devel] Proxying SIP live RTP to RTSP problem (segfault)

2017-09-24 Thread Ross Finlayson
> While the H264LiveServerMediaSession constructor is called once - the > createNewRTPSink/createNewStreamSource are called twice (?!) upon RTSP > connection. May be this is the real reason of > "FramedSource[0x1bf42e0]::getNextFrame(): > attempting to read more" message ? No, I don’t think so, b

Re: [Live-devel] Proxying SIP live RTP to RTSP problem (segfault)

2017-09-24 Thread rus
> Aha - you should be able to overcome this by calling: > subsession->initiate(); > just before your call to > createRTSPServer(subsession); Well, this gives us the previous message (please note that -r flag is still passed to the playSIP) : "FramedSource[0x1bf42e0]::getNextFrame():

Re: [Live-devel] Proxying SIP live RTP to RTSP problem (segfault)

2017-09-24 Thread Ross Finlayson
> Well, running playSIP with -r flag gives just the another SIGSEGV without any > error messages: Aha - you should be able to overcome this by calling: subsession->initiate(); just before your call to createRTSPServer(subsession); > >> I should also point out, however, that the

Re: [Live-devel] Proxying SIP live RTP to RTSP problem (segfault)

2017-09-24 Thread rus
Hi, > The key to understanding your problem is this line: > > "FramedSource[0x2493050]::getNextFrame(): attempting to read more than once > > at the same time!? > What?s happening (according to the stack trace) is that the ?FramedSource? > object that?s being fed into the ?H264VideoStreamDiscre

Re: [Live-devel] Proxying SIP live RTP to RTSP problem (segfault)

2017-09-24 Thread Ross Finlayson
The key to understanding your problem is this line: > "FramedSource[0x2493050]::getNextFrame(): attempting to read more than once > at the same time!” What’s happening (according to the stack trace) is that the “FramedSource” object that’s being fed into the “H264VideoStreamDiscreteFramer” obje

[Live-devel] Proxying SIP live RTP to RTSP problem (segfault)

2017-09-24 Thread rus
Greetings, I need to to dynamically create the RTSP endpoints from the live SIP calls (video stream only). So first I've tested the playSIP with Asterisk (small patch to liveMedia/SIPClient.cpp needed) and successfully got the raw out.264 file recorded with the video from the call. Next I've mod