Yes i guess you could use groups for phones, but what if you have say 3 sites, each with their own voice switch? Site 1 and site 2 have 10Mb/S lines, but site 3 has 1Mb/s and the users are also using terminal services. They only want to commit 200k to voice, but they want 6 call capacity. You are now talking sipX to sipX and unless all the phones are set to only use a low bandwidth codec, even for internal calls, you are going to get a high bandwidth call.
Perhaps the sip trunks could be enhanced with codec lists. When you create a trunk between two sites, the SIP trunk will pass on only the allowed codecs for negotiation. If the calling endpoint does not offer a suitable codec, then a "488 not acceptable" would be returned from the SBC? Maybe I am getting too ambitious. -- Regards Mark Dutton _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
