Hi,

I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 +
opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts.

Firefox:
Neither sipml5 or jssip works with calls to asterisk, audio/video doesn’t matter. WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684 RTP/SAVPF 109 0 8 101
–> Asterisk sends “SIP/2.0 488 Not acceptable here”

Chrome:
I’ve tried both sipml5 and jssip softphones and they both work. Even video + confbridge works with some minor quirks (lost connections sometimes, I guess plain old nat issues). Just relaying audio+video with confbridge to a handful of participants seems to use quite a bit of cpu thought.

Screen-share:
This works, but Confbridge is not very happy about a channel with video
(vp8) and not audio and is printing this 80 times a second:

WARNING[8919][C-00000000] channel.c: Unable to find a codec translation path from (vp8) to (slin)
WARNING[8919][C-00000000] chan_sip.c: Asked to transmit frame type slin, while native formats is (vp8) read/write = unknown/unknown WARNING[8919][C-00000000] channel.c: Don’t know any of (vp8) formats

How do you think about adding webrtc to a existing Asterisk/Kamailio environment? Do you use kamailio (websockets) as a front, a dedicated webrtc asterisk or something like webrtc2sip?

How do you use / plan to implement webrtc in your environment?

Any feedback is welcome. Thanks!