<br><br><div class="gmail_quote">2010/3/17 Vinícius Fontes <span dir="ltr"><<a href="mailto:vinicius@canall.com.br">vinicius@canall.com.br</a>></span><br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">
----- "Kevin Sandy" <<a href="mailto:kevin.sandy@snohio.net">kevin.sandy@snohio.net</a>> escreveu:<br>
<div><div></div><div class="h5"><br>
> We're having an odd issue with codec negotiation from one of our SIP<br>
> providers. Here's the basic situation.<br>
><br>
> We receive an invite from them advertising support for G711, G729, and<br>
> G723. In our response, we send back that we support G711 and G729. In<br>
> about half the cases, this results in no problems, with audio being<br>
> encoded with G711. The other half of the time, they send us a second<br>
> invite requesting G729. However, they proceed to send us a G711<br>
> encoded audio stream...<br>
><br>
> They have somewhat acknowledged the problem, but their advice is for<br>
> us to only accept a single codec in our 200 OK. We don't want to<br>
> disable either; we have customers using G729, so we'd like to avoid<br>
> transcoding when possible, but we also do some T38 faxing, which I<br>
> believe requires G711 to start off.<br>
><br>
> My first thought was to selectively force the codec on inbound calls -<br>
> if it is for a voice number, use 729, otherwise 711. However, I can't<br>
> find any way of doing this within Asterisk. (We do have an OpenSIPS<br>
> server sitting between us and the provider, and I could use OpenSIPS<br>
> features to do this; however, right now the OpenSIPS server is fairly<br>
> dumb - it's only proxying traffic between us and the provider and<br>
> knows nothing about our specific DIDs.)<br>
><br>
> A couple more details in case anyone has seen a similar issue. The<br>
> provider is Broadvox, and this issue only seems to manifest on calls<br>
> coming to them via Skype. They claim to not have any direct link with<br>
> Skype, but it seems odd that the problem would be specific to Skype<br>
> callers if the call is coming to Broadvox as a standard PSTN call.<br>
><br>
> Is there any way to do this? Am I totally missing something and making<br>
> a stupid mistake, or making the issue more complicated than it needs<br>
> to be?<br>
><br>
<br>
</div></div>If your only concern about using G711 is regarding T38, go ahead and enable G729 only. T38 doesn't need G711 at all.<br>
<div><div></div><div class="h5"><br></div></div></blockquote><div><br>If your customers don't mind G729 then what is said above is fine. <br><br>There will be a T.38 reinvite so it won't be G729 anymore. Canreinvite does not need to be set to yes for this to work in your sip.conf either. It can be confusing but they are different types of reinvites.<br>
<br>Thanks,<br>Steve Totaro.</div></div>