Steve- > 2010/3/17 VinÃcius Fontes <[email protected]> > >> ----- "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. >> >> > 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 -- _____________________________________________________________________ -- 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
