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. :) I have a few more questions, but I'm guessing that I can figure them out on my own once I have the code. -- _____________________________________________________________________ -- 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