asterisk users - Sep 2015 - Fw: Issue with audio: Local Asterisk + WebRTC

Reply to: asadovalros at gmail.com





Hello everyone. I'd appreciate a lot your help with this issue. I'm
running a very basic script of JS for subscribing my jsSIP User Agent to my
local Asterisk server and making a voice call. I don't get any warnings or
errors from the Asterisk CLI nor the script, but when I make a call to a legacy
SIP phone or SIP trunk well configured, there is no audio on any side although
there is ringing, calls can be answered and they never drop. My Asterisk 12 was
compiled with SRTP and pjproject.




I read at the Asterisk WebRTC
Wiki(https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support) this:
"Starting with Asterisk 12 you need to have pjproject libraries installed,
otherwise you most likely won't have audio in your WebRTC calls and no
warning whatsoever!"
I properly installed it and selected it for the Asterisk compilation, but I
wonder wether I did it wrong, and how can I check it ...



I leave here my Asterisk files: http://pastebin.com/p5euwnTJ

This is a SIP debuging of my jsSIP UA subscribing: http://pastebin.com/KxgB6GYb

This is a SIP debugging of a local call: http://pastebin.com/VQayVYAh

Finally this is what the CLI says about it: http://pastebin.com/9FXAUU6c


... Thanks in advance
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On Tue, Sep 8, 2015 at 8:28 AM,  <asandovalros at gmail.com>
wrote:
> Reply to: asadovalros at gmail.com
>
> Hello everyone. I'd appreciate a lot your help with this issue. I'm
running
> a very basic script of JS for subscribing my jsSIP User Agent to my local
> Asterisk server and making a voice call. I don't get any warnings or
errors
> from the Asterisk CLI nor the script, but when I make a call to a legacy
SIP
> phone or SIP trunk well configured, there is no audio on any side although
> there is ringing, calls can be answered and they never drop. My Asterisk 12
> was compiled with SRTP and pjproject.
>
> I read at the Asterisk WebRTC
> Wiki(https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support)
> this: "Starting with Asterisk 12 you need to have pjproject libraries
> installed, otherwise you most likely won't have audio in your WebRTC
calls
> and no warning whatsoever!"
> I properly installed it and selected it for the Asterisk compilation, but I
> wonder wether I did it wrong, and how can I check it ...
Well, to make sure pjsip was installed correctly you could make a
basic PJSIP to PJSIP (without websockets involved) call between
phones. However I recommend you upgrade to Asterisk 13 and try again..

Asterisk 12 went into Security fix only on 2014-12-20 , since that
date it has not received any non-security bug fixes. With WebRTC being
fairly "bleeding edge" you'll want to use Asterisk 13 which will
have
any available updates required to work with the browsers.

Other users will also be much more eager to help support you on a
currently supported Asterisk version.

https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions

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Rusty Newton
Digium, Inc. | Community Support Manager
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direct: +1 256 428 6200

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