[asterisk-users] pjsip: asterisk can't decide which codec to use

Joshua Colp jcolp at digium.com
Fri May 12 13:49:40 CDT 2017

On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote:


> If I'm doing exactly the same call originated with another extension,
> there can't be seen these frequent changes. But the strange thing is,
> that in both cases the part between extension and asterisk doesn't show
> any codec changes ... .
> Deeper investigations show, that if the conference (callee) sends the
> first rtp package (-> g711 - should be g722), things are going choppy, 
> if the extension (caller) sends the first package (g722), things are 
> running stable.
> Any idea to convince asterisk always to use the first codec of ok sdp 
> or how to convince asterisk to put only one codec to ok sdp (the first).

This is not currently an option in chan_pjsip but I'd suggest filing an
issue[1] for this scenario with all available information.

[1] https://issues.asterisk.org/jira

Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

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