Hello,
I was able to use webrtc2sip and connect audio calls in g729 passthrough
and ulaw modes over a callus webpage js.
However not tested Video.
and it worked good even on AST 1.8.XX
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID: +91-22-71967196
Cell: +91-9820332422
On Mon, Apr 14, 2014 at 2:26 PM, Johan Wilfer li...@jttech.se wrote:
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-0005] 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-] channel.c: Unable to find a codec translation
path from (vp8) to (slin)
WARNING[8919][C-] chan_sip.c: Asked to transmit frame type slin,
while native formats is (vp8) read/write = unknown/unknown
WARNING[8919][C-] 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!
--
Johan Wilfer
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