On 29 Jan 2015, at 23:56, Muhammad Shahzad <shaherya...@gmail.com> wrote:
> Hi, > > This may be a bit out of focus topic for this forum but i am posting it here > anyway with hope that some guru would shed some light on it and point me to > right direction. > > The problem is that i want to establish video call between a webrtc and a sip > client using kamailio (for signalling) and RTPEngine (for media relay). Both > signalling and the audio stream seems to work perfectly fine The remote video > on webrtc client side (i.e. video stream from sip client) takes about 20-30 > seconds to establish but once it starts it works fine. However, the remote > video on sip client side (i.e. video stream from webrtc client) starts almost > immediately (within 3-5 seconds) but it gets stuck after 1 or 2 seconds, then > it goes blank after about 30 seconds. > > After a long discussion with sip client developer, we now understand the fact > that sip client sends a request for so called key-frame, which is ignored by > webrtc client. This request is sent through both RTCP stream and SIP INFO > message. > > The SIP INFO message seems to be pointless as media is internally managed by > chrome/firefox and these browsers don't give us such sophisticated access and > control over media streams. Please let me know if this assumption is wrong. > > For the RTCP stream based request (RTCP-FIR), i only see "Invalid RTCP packet > type" error message in RTPEngine logs (not sure if it drops this packet or > relay it anyway). > > Does anyone has any idea on how can we either, > > 1. Force WebRTC client (running on Chrome / Firefox) to honor SIP INFO > message and issue a key-frame in RTP video stream in response to this SIP > request? Talk with the SIP stack developer. I don't know if it's possible at all and I think using SIP info for this is more or less the old way. Sending it in the actual media stream feels like a more modern and better way. > > OR > > 2. Force RTPEngine to accept RTCP-FIR and issue key-frame in RTP video stream > on webrtc client's behalf? File a bug report with the RTPengine team. It's clearly something they need to support. /O > > If there is any other solution to this, please feel free to share. > > > Thank you. > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users@lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
