WipeOut wrote: > Here is the scenario... > > SIP UA's can use either GSM or G.711 ( in that order of preference in > the sip.conf ).. > Asterisk Server1 is linked to Asterisk Server2 via IAX2 and also > supports GSM and G.711 ( also in that order of preference).. > > 1. If a call comes in from the UA using GSM and then goes out over the > IAX2 leg, Will Asterisk simply move the GSM encoded data from the SIP > channel to the IAX2 channel and so have very little performance > overhead > since it will not be decoding and re-encoding the GSM data? or will it > decode and re-encode the voice data? > > 2. Assuming that the answer to question1 is that Asterisk will simply > move the encoded data from the SIP channel to the IAX2 channel without > decoding and re-encoding (otherwise the question is irrelevant), > would a > call coming in in SIP using G.711 be converted to GSM on the IAX2 > channel since it is higher in the order of preference or will it > identify that G.711 is also available on the IAX2 leg and so switch > the > call using G.711 to save the performance overhead so that the data > does > not have to be decoded and re-encoded? > > Thanks.. > > Later.. > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Hi, I had a same thought few days ago. Also, I would like to add/ask: If a call comes to Server1 by SIP, is it possible to re-direct client to Server2. In another words, IAX2 part is (taken out), so client communicates with Server2 by SIP directly during that call. My primary motivation behind this is to save on resources. Ta SJ _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
