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?


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