asterisk users - Apr 2014 - Webrtc and adventures with Asterisk 11

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!

-- 
Johan Wilfer

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: mitul at enterux.in
DID: +91-22-71967196
Cell: +91-9820332422



On Mon, Apr 14, 2014 at 2:26 PM, Johan Wilfer <lists at 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-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!
>
> --
> Johan Wilfer
>
>
> --
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