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

[Michael Maier]

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

https://issues.asterisk.org/jira/browse/ASTERISK-26996


Thanks,
regards,
Michael