[asterisk-users] SIP codec negotiation / manipulation

[Olle E. Johansson]

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