You just reminded me now, our ITSP can force SIP endpoints (phones) to use certain codec by way of reordering the priority of codecs in the SDP. This way you configure the phones to use these codecs in this order alaw, ulaw and g729. This should allow phones on the LAN to use alaw (first choice). When any of the phones make outbound call using the SIP trunk, ITSP will reorder the codecs by putting g729 on top of the list and since phones are capable of negotiating the use of g729 voila.
Would that work? Have to test it with our provider, may have to request them to prefer G729 (put it on top of list) first. -----Original Message----- From: M. Ranganathan [mailto:[email protected]] Sent: Saturday, 1 August 2009 9:07 a.m. To: Yakout Esmat Cc: [email protected] Subject: Re: [sipx-users] CODEC scenario On Fri, Jul 31, 2009 at 5:02 PM, Yakout Esmat<[email protected]> wrote: > As Paul mentioned. You would want to consume the least amount of bandwidth > per call over your choked up DSL link, while on the LAN you want to use the > better quality G711 CODEC. > > The way Cisco does it is, group extensions that are on the same LAN together > in a logical group and then specify that all extensions within this group > use alaw, while communication between this group and other groups > (supposedly over the WAN) use G729. > > The point here is, by grouping extensions together you can specify which > CODED(s) could be used Intra-group and which ones could be used Inter-group. > This way give admins much needed flexibility. > > By using the group concept you can also configure many other parameters that > only relates to intra-group conversations. > > cheers One way this can be accomplished is by editing the SDP on the way to the ITSP. All IN/OUT signaling activity for a given ITSP that has a codec restriction, gets the filter applied. I like using the notion of an exclude set better than an allow set as suggested by Paul -- there is less work for sipxbridge to do. > > > > -----Original Message----- > From: M. Ranganathan [mailto:[email protected]] > Sent: Saturday, 1 August 2009 7:42 a.m. > To: Yakout Esmat > Cc: [email protected] > Subject: Re: [sipx-users] CODEC scenario > > On Fri, Jul 31, 2009 at 3:16 PM, Yakout Esmat<[email protected]> wrote: >> Hi, >> >> >> >> We all know that sipX doesn't transcode and it doesn't need to which is >> great but in saying that can we set up a scenario like this one: >> >> >> >> A sipX 4.0 server and a couple of Aastra phones on the LAN (Aastra phones >> are capable of doing G711 and G729 ) >> >> >> >> 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. > > Ranga > > >> >> >> >> Thanks >> >> >> >> _______________________________________________ >> 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/ >> > > > > -- > M. Ranganathan > > > -- M. Ranganathan _______________________________________________ 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/
