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[asterisk-users] Webrtc and adventures with Asterisk 11


 
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lists at jttech.se
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PostPosted: Mon Apr 14, 2014 3:56 am    Post subject: [asterisk-users] Webrtc and adventures with Asterisk 11 Reply with quote

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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mitul at enterux.in
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PostPosted: Mon Apr 14, 2014 4:09 am    Post subject: [asterisk-users] Webrtc and adventures with Asterisk 11 Reply with quote

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@enterux.in (mitul@enterux.in)
DID: +91-22-71967196
Cell: +91-9820332422





On Mon, Apr 14, 2014 at 2:26 PM, Johan Wilfer <lists@jttech.se (lists@jttech.se)> wrote:
Quote:
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


--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
              http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users
Back to top
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