[asterisk-users] SIP codec negotiation / manipulation

[Vinícius Fontes]

----- "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.