On Fri, 2009-07-31 at 15:55 -0400, Paul Mossman wrote: > > Ranga wrote: > ... > > > Can we set these up in a way that forces the phones to use alaw > > > (g711) between each other over the LAN and use G729 over > > the SIP trunk > > > (SIP trunk over DSL) to provider? > > > > > > No you cannot do that. May I ask you why you want to do that? > > We used to have codec filtering in sipxbridge but it is an > > ugly hack and does not exist any longer for good reason. > > Codecs are negotiated end to end in sipx. > > Because he wants calls out over his DSL line to chew as little bandwidth > as possible. The only way we can do it today is to configure the phones > to use only G.729. But then local calls wouldn't be using the much > higher bandwidth available on the LAN. > > It would be nice to be able to configure, for each SIP Trunk, the list > of codecs may be used. i.e. All SDP offers that pass through sipXbridge > would have disallowed codecs removed.
The trouble is that at the time you send the INVITE (from the phone), you don't have any way of knowing where the call will go. If I tell my extension to ring both my desk phone (LAN) and my cell phone (PSTN, and therefor possibly that DSL line), it's the _same_ invite forked at the proxy that goes to both. _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
