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

Michael Maier m1278468 at mailbox.org
Sat May 13 00:21:52 CDT 2017

On 05/12/2017 at 08:49 PM, Joshua Colp wrote:
> On Fri, May 12, 2017, at 02:46 PM, Michael Maier wrote:
> <snip>
>> 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



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