Re: PJSIP Codec Negotiation Issue

"Joshua C. Colp" <[email protected]>
Newsgroups gmane.comp.telephony.pbx.asterisk.user
Message-ID <CAM0A2Z0z8TxcpcZsHOcG3W1nxrEJGboGeybf4juRxsE0hsvN9g@mail.gmail.com>
On Fri, May 20, 2022 at 11:30 AM Benoît Panizzon <[email protected]>
wrote:

> Hi Joshua
>
> > What is the specific issue that is happening? If it's that one call
> > leg negotiated at opus and the other at alaw, that is currently the
> > way things still work. Each call leg is still ultimately negotiated
> > independently so the A leg can be opus, and the B leg can be alaw. I
> > hope that we're able to eventually return to codec negotiation work
> > to improve that with the foundation put into place previously, but I
> > don't know when that will happen.
>
> It looks like I had some other misconfiguration also causing some weird
> effects.
>
> Well wheat I wonder is how codec negotiation should work in this
> situation.
>
> I am aware, the SDP defines what the device can send, not what it is
> able to receive. But can't we assume this is the same?
>
> Device A <=> Asterisk <=> TSP Trunk <=> [WORLD] <=> Device B
>
> Let's assume Device B is alaw only, but we don't know, it could also be
> possible of better quality codecs.
>
> Device A support opus,g711,alaw
>
> So towards Device A I would define !all,opus,g722,alaw
> And towards the TSP Trunk the same.
>
> But when Device B replies with an SDP containing 'only' alaw, shouldn't
> this SDP be passed to Device A which then hopefully use the same codec?
>

Asterisk isn't a SIP proxy, so there is no passing of SDP. Each call leg is
currently independent on the answer. So between Device A and Asterisk is
negotiated separately from Asterisk and Device B. Each side can end up
using different codecs, and Asterisk would transcode if possible.


>
> Of course Device A can receive alaw and send opus. But device B might
> not understand opus.
>
> And I just realized, this advanced codec negotiation is only available
> on Asterisk 18. I'm still on 16 :-)
>


-- 
Joshua C. Colp
Asterisk Technical Lead
Sangoma Technologies
Check us out at www.sangoma.com and www.asterisk.org

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
      https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
lmpx.com only provides a reader for public news (NNTP) servers. It is not affiliated with the servers or forums shown here and is not responsible for the content of articles, which is written by their respective authors.