[asterisk-users] SIP codec negotiation / manipulation

[Jeff Brower]

Steve-


> On Wed, Mar 17, 2010 at 6:02 PM, Jeff Brower <jbrower at signalogic.com> wrote:
>
>      Steve-
>
>      > 2010/3/17 Vin¨ªcius Fontes <vinicius at canall.com.br>
>      >
>      >> ----- "Kevin Sandy" <kevin.sandy at snohio.net> escreveu:
>      >>
>      >> > 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?
>      >> >
>      >>
>      >> If your only concern about using G711 is regarding T38, go ahead and
>      enable
>      >> G729 only. T38 doesn't need G711 at all.
>      >>
>      >>
>      > If your customers don't mind G729 then what is said above is fine.
>      >
>      > 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.
>
>      I don't see how this can work if Broadvox then sends G711 anyway.  I
>      understand that to be the OP's root problem.
>
>      -Jeff
>
>
> It doesn't matter what the codec is initially, if the provider supports T.38 and
> you do too, a reinvite is sent changing whatever codec over to T.38.

I meant for the Broadvox voice output, but maybe your suggestion works Ok and solves
his problem.

-Jeff

-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100317/f492bcb9/attachment.htm