17 mar 2010 kl. 16.37 skrev Kevin Sandy: > 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? > The problem here is that you have a proxy in between, so Asterisk can't have separate peer configurations, since all the SIP messages are from the same IP and thus the same peer. I have a branch that implements peer matching in this specific configuration, which means that you can have different codec configurations for different partners even though there's a proxy in front of Asterisk.
https://origsvn.digium.com/svn/asterisk/team/oej/pinetree-1.4 Please try this branch and give feedback. There should be some docs in sip.conf for the new "matchrule" setting. /O -- _____________________________________________________________________ -- 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
