Hi all,
we are evaluating WCS 5.1 (5.1.3688) as SIP to RTMP GW (publishing towards wowza a videoconference established via tandberg (cisco)).
As far as I read they are compatible, aren't they?
We set up the call via REST API: Sip session is established, RTP audio (g711) and video (h264) are received from Cisco and streamed via RTMP towards Wowza, but no RTP seems to be originated from WCS to Cisco (even if the SDP exchange seems to be sendrecv, Sip script attached, sip headers removed for privacy). This seems to cause Cisco to close UDP ports before we can manage to send the dtmf code needed to join the videocall.
I would expect some RTP from WCS automatically: isn't it so? I'm wondering whether I'm missing some configuration property (we attach the flashphoner.properties file) or where do I go wrong...
Some NPEs show up in the server logs, I do not know if they are meaningful or not:
4:42:44,433 INFO SipUserAgent - HTTP-pool-2-thread-1 NPE DEBUG callRequest: INVITE sip:* SIP/2.0
14:42:44,477 INFO SipCall - HTTP-pool-2-thread-1 NPE debug sipMediaSession MediaSession{remoteHostAddress='null', isControlling=true, initialized=false, id='provaDTMF_127.0.0.1:-8592093358937445012', agent=null, audioSession=Session{secured=false, terminated=false, localPort=31002, rtcpMux=false, sdpState=sendrecv}, videoSession=Session{secured=false, terminated=false, localPort=31004, rtcpMux=false, sdpState=inactive}} audioDescriptionConfig DescriptionConfig{proxySession=Session{secured=false, terminated=false, localPort=31006, rtcpMux=false, sdpState=sendrecv}, isControlling=true, sdpState=sendrecv} videoDescriptionConfig DescriptionConfig{proxySession=Session{secured=false, terminated=false, localPort=31008, rtcpMux=false, sdpState=inactive}, isControlling=true, sdpState=sendrecv}
Thanks for any help you could provide me
Best Regards
Silvia
we are evaluating WCS 5.1 (5.1.3688) as SIP to RTMP GW (publishing towards wowza a videoconference established via tandberg (cisco)).
As far as I read they are compatible, aren't they?
We set up the call via REST API: Sip session is established, RTP audio (g711) and video (h264) are received from Cisco and streamed via RTMP towards Wowza, but no RTP seems to be originated from WCS to Cisco (even if the SDP exchange seems to be sendrecv, Sip script attached, sip headers removed for privacy). This seems to cause Cisco to close UDP ports before we can manage to send the dtmf code needed to join the videocall.
I would expect some RTP from WCS automatically: isn't it so? I'm wondering whether I'm missing some configuration property (we attach the flashphoner.properties file) or where do I go wrong...
Some NPEs show up in the server logs, I do not know if they are meaningful or not:
4:42:44,433 INFO SipUserAgent - HTTP-pool-2-thread-1 NPE DEBUG callRequest: INVITE sip:* SIP/2.0
14:42:44,477 INFO SipCall - HTTP-pool-2-thread-1 NPE debug sipMediaSession MediaSession{remoteHostAddress='null', isControlling=true, initialized=false, id='provaDTMF_127.0.0.1:-8592093358937445012', agent=null, audioSession=Session{secured=false, terminated=false, localPort=31002, rtcpMux=false, sdpState=sendrecv}, videoSession=Session{secured=false, terminated=false, localPort=31004, rtcpMux=false, sdpState=inactive}} audioDescriptionConfig DescriptionConfig{proxySession=Session{secured=false, terminated=false, localPort=31006, rtcpMux=false, sdpState=sendrecv}, isControlling=true, sdpState=sendrecv} videoDescriptionConfig DescriptionConfig{proxySession=Session{secured=false, terminated=false, localPort=31008, rtcpMux=false, sdpState=inactive}, isControlling=true, sdpState=sendrecv}
Thanks for any help you could provide me
Best Regards
Silvia
