----- "Kevin Sandy" <[email protected]> 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. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
