[asterisk-users] Inquiry:Asterisk different codec schemes?

hadi motamedi motamedi24 at gmail.com
Mon Jan 4 22:45:52 CST 2010


On Mon, Jan 4, 2010 at 1:46 PM, Kevin P. Fleming <kpfleming at digium.com>wrote:

> hadi motamedi wrote:
>
> > Sorry . I didn't get the point clearly . In the SIP Invite message , it
> > says "my audio endpoint is IP x.x.x.x port x, and I can use codecs
> > A,B,C". The remote endpoint responds with a 200 OK, saying "my audio
> > stream is at IP y.y.y.y port y, and I choose codec B". Can you please do
> > me favor and let me know if my understanding is right or not ?
> > Thank you
>
> No, you are not understanding the SDP offer/answer model properly. If
> one endpoint offers codecs A, B and C in its SDP, it is willing to
> *receive* media in those formats. The receiver of that offer can choose
> to send media to the offerer in any of those formats, at any time. If
> the answering endpoint includes only codec B in its SDP, then it is
> willing to *receive* only codec B. In that scenario, it is possible for
> media to flow from endpoint 1 to endpoint 2 using codec B, and from
> endpoint 2 to endpoint 1 using codec A (or C), but this will not happen
> if Asterisk is an endpoint in this scenario.
>
> When Asterisk receives a media frame, if the format of that frame is not
> the format that it is currently sending to the other endpoint, it will
> switch to that format automatically. If it cannot do so because the
> other endpoint did not offer to receive that format, then the call's
> audio will probably fail. This is the reason why I responded before that
> Asterisk does not support asymmetric formats in a media session.
>
> In reality, it is extremely uncommon for a SIP endpoint to want to send
> media in a format that it is not also willing to receive; in fact, I
> can't say I've ever seen this situation arise in any testing I've done
> or in any issues reported in our issue tracker.
>
> --
>  Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kpfleming at digium.com
> Check us out at www.digium.com & www.asterisk.org
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>


Thank you very much for correcting me .
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100105/3024c76f/attachment.htm 


More information about the asterisk-users mailing list