[asterisk-users] Fw: Issue with audio: Local Asterisk + WebRTC

[Rusty Newton]

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

-- 
Rusty Newton
Digium, Inc. | Community Support Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct: +1 256 428 6200

Check us out at: http://digium.com & http://asterisk.org