If you're using a conference bridge, then you are using a G.729 license for each participant that is using G.729. The reason is that FreeSWITCH has to multiplex all the audio together and the only way to do this is to transcode the audio to a standard codec, mix the audio, and reencode the stream back to G.729.
On Tue, Nov 15, 2011 at 10:00 AM, Domenico Chierico <[email protected]>wrote: > On Tue, 2011-11-15 at 09:44 -0500, Tony Graziano wrote: > > First of all realize the g729 codec passthru and full g729 codec is no > > the same. Passthru means one way (to voicemail or to an endpoint that > > supports it). There are issues with g729 getting prompts properly > > transcoded, etc. > > > I'm using full g729 for conference. > > > > Also realize there is a wiki page about perhaps using the g720 license > > for freeswitch, but I had not ventured there for some time: > > > > > > http://wiki.sipfoundry.org/display/sipXecs/G729 > > it's work quite well, I haven't problems setting up call to the > conference with g729 comm codec > > > > > > The 4.2 version was running at one point an incompatible FS engine > > after the FS license guys changed up their binary and engine > > requirements. It might work, and it might not. If you go this > > direction, please ensure if it works you update the wiki with the > > changes needed (if any) to make this functional. > > > I'm working on 4.4 > > > > Also realize when you TRANSCODE you will be consuming some CPU and RAM > > and might want to take that into consideration for sizing. The other > > option is to use an SBC which transcodes G729 for you. > > > > > why i should consider transcoding if all the phone and other elements > works in g729? > > myabe I was unclear 'cose of my poor english .. so sorry me. > > > > _______________________________________________ > sipx-dev mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-dev/ > -- Josh Patten eZuce Solutions Architect O.978-296-1005 X2050 M.979-574-5699
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