22 mar 2010 kl. 14.54 skrev Kevin Sandy: > > > On 3/21/2010 4:05 AM, Olle E. Johansson wrote: >> > >> 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 > > > I'd be interested in trying this out - but the site doesn't seem to be > responding. :) Sorry, gave you the developer URL. Too quick copy and paste... Here's a correct one: >> http://svn.digium.com/svn/asterisk/team/oej/pinetree-1.4
/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