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

[Michael Maier]

On 05/13/2017 at 07:21 AM Michael Maier wrote:
> 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

I just tested your fix 2 times w/ using the scenario mentioned in the
bug report. It has been working for me. No more flipping.

Asterisks indeed commits more than one codec in ok sdp, but always uses
the first one afterwards. Hopefully the peer always handles it the same
way. I would have thought that the ok sdp contains just one codec (the
best).


Thanks,
Michael