[asterisk-users] SIP codec negotiation / manipulation

Jeff Brower jbrower at signalogic.com
Wed Mar 17 17:02:10 CDT 2010


Steve-

> 2010/3/17 Vinícius Fontes <vinicius at canall.com.br>
>
>> ----- "Kevin Sandy" <kevin.sandy at snohio.net> escreveu:
>>
>> > We're having an odd issue with codec negotiation from one of our SIP
>> > providers. Here's the basic situation.
>> >
>> > We receive an invite from them advertising support for G711, G729, and
>> > G723. In our response, we send back that we support G711 and G729. In
>> > about half the cases, this results in no problems, with audio being
>> > encoded with G711. The other half of the time, they send us a second
>> > invite requesting G729. However, they proceed to send us a G711
>> > encoded audio stream...
>> >
>> > They have somewhat acknowledged the problem, but their advice is for
>> > us to only accept a single codec in our 200 OK. We don't want to
>> > disable either; we have customers using G729, so we'd like to avoid
>> > transcoding when possible, but we also do some T38 faxing, which I
>> > believe requires G711 to start off.
>> >
>> > My first thought was to selectively force the codec on inbound calls -
>> > if it is for a voice number, use 729, otherwise 711. However, I can't
>> > find any way of doing this within Asterisk. (We do have an OpenSIPS
>> > server sitting between us and the provider, and I could use OpenSIPS
>> > features to do this; however, right now the OpenSIPS server is fairly
>> > dumb - it's only proxying traffic between us and the provider and
>> > knows nothing about our specific DIDs.)
>> >
>> > A couple more details in case anyone has seen a similar issue. The
>> > provider is Broadvox, and this issue only seems to manifest on calls
>> > coming to them via Skype. They claim to not have any direct link with
>> > Skype, but it seems odd that the problem would be specific to Skype
>> > callers if the call is coming to Broadvox as a standard PSTN call.
>> >
>> > Is there any way to do this? Am I totally missing something and making
>> > a stupid mistake, or making the issue more complicated than it needs
>> > to be?
>> >
>>
>> If your only concern about using G711 is regarding T38, go ahead and enable
>> G729 only. T38 doesn't need G711 at all.
>>
>>
> If your customers don't mind G729 then what is said above is fine.
>
> There will be a T.38 reinvite so it won't be G729 anymore.  Canreinvite does
> not need to be set to yes for this to work in your sip.conf either.  It can
> be confusing but they are different types of reinvites.

I don't see how this can work if Broadvox then sends G711 anyway.  I understand that to be the OP's root problem.

-Jeff




More information about the asterisk-users mailing list