Hi,
rtpproxy and mediaproxy do media-relaying and OpenSER does SIP routing. Both have nothing to do with codecs, which are negotiated end-to-end. So yes, OpenSER and rtpproxy/mediaproxy do support G729 in a sense that they allow UACs to communicate with G729. If you want to transcode between codecs (e.g. G711 <-> G729), you need - as Daniel already said - a transcoder like Asterisk.
Andreas
raviprakash sunkara wrote:
Hello Daniel ,
I'm Also Having the Same doubt on g729 Codec,
I'm using RTPproxy with Nathelper , OpenSER and RTP proxy does media signaling when the Call is Established,
My main question is Is RTP proxy support the G729, with OpenSER,
With out using the Transcoder ( Asterisk ) How can OpenSER signals the G729 Codec.
On 2/15/07, *Daniel-Constantin Mierla * <daniel@voice-system.ro mailto:daniel@voice-system.ro> wrote:
Hello, you need a transcoder in the middle. OpenSER does only signaling, so it is not able to transcode. Asterisk, for example, does. Cheers, Daniel On 02/15/07 10:57, tusker keg wrote: > Howdy > > I have a situation I hope you guys will help me out with > > I am receiving call from a VOIP peer (SIP Call) and the peer can > only send them as G711. > I need to redirect to call to another voip peer over the wan and due > to bandwidth considerations I need to translate the codec to g729. > > Any ideas on how to do this, > > Sample config file will be help > > Regards > > ./Tusker > > _______________________________________________ > Users mailing list > Users@openser.org <mailto:Users@openser.org> > http://openser.org/cgi-bin/mailman/listinfo/users > _______________________________________________ Users mailing list Users@openser.org <mailto:Users@openser.org> http://openser.org/cgi-bin/mailman/listinfo/users
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