I have been trying to configure asterisk to do webrtc call.
Here are the system details:
Asterisk Version : Asterisk 13.5.0.
OS version : Linux ubuntu 3.16.0-30-generic
Browser : Chrome Version 45.0.2454.101 (64-bit)
Problem : I could make audio call work between SIPML5 and a sip phone. (Xlite, Eyebeam)
But when I make a video call, call gets established but no video on both sides. Only audio is heard.
User details in sip.conf. (Assuming everything other configuration is ok as audio call is working fine)
allow = ulaw,h264,vp8
disallow = all
allow = ulaw,h263,h264
Debugging information. (I took some traces using wireshark)
- Call is initiated by webrtc client.
2.Sdp exchange between sip phone and webrtc happens correctly. SIP phone receives ulaw/H263,H264
in INVITE and responds with ulaw/H263.
- Webrtc side sends invite with ulaw/VP8 and receives ulaw/VP8 in 200OK from asterisk.
- When I looked at the RTP packets on sip phone side, The payload is dynamic and when I set filter
in wireshark to interpret the payload 100 as vp8, I see vp8 packets coming from asterisk.
(This I see as a problem. Not sure why it is not sending H263(a static payload type 34) and sending
VP8. Doesnt asterisk do Video transcoding? If no, shouldnt it reject the call instead of sending
proper sdp to sip side?)
- I also tried this on Firefox. Assuming that if phone and browser both can do H264, it should work.
But no luck there too. Unfortunately I did not take any trace there.
- Even video between 2 SIPML extensions does not happen. Though call gets established with audio.
- I have tried setting directmedia to yes/no.
Am I missing some configuration on Asterisk side for video?
Any kind of debugging help on this will be a great thing.