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..
If the incoming and outgoing Codecs are the same, there is no "conversion" done. It basically becomes a packet relay, what goes in, comes out. I'm not sure of the answer to your second question. However, your question actually begs a question I've been wondering about in the last couple of days: I'm doing H.323 in, H.323 out....simple relay. (This is my customer's requirement...not my preference). What I want to do is ALWAYS use the same codec for the outgoing leg as for the incoming leg. In other words, if the call comes in as G.729, the outgoing call uses G.729 ONLY. If the incoming call is G.711, I want the outgoing to be G.711. I want to avoid any sort of transcoding. Is it possible? Thanks. Tom Lowe (FYI, Dual Xeon 3.06, 120 channels (60 calls) of above scenario, G.729 using less than 10% CPU!) (Remember, no transcoding is being performed) -----Original Message----- From: WipeOut [mailto:wipe_out@users.sourceforge.net] Sent: Monday, January 12, 2004 6:54 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] A question on codec translation. 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-usersHi, 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