Irakli Natsvlishvili
2005-Sep-07 00:04 UTC
[Asterisk-Users] asterisk, SIP, Re-INVITEs and different contexts
Hmmm... Folks, I beg you pardon, if I'm telling something which was said before, but actually I have not found this anywhere, neither on Voip-info.org or in several Asterisk's docs. So, here is the statement: If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them will ALWAYS go via Asterisk. I.e. Asterisk WILL NOT issue Re-INVITE even if: 1. Both UAs have canreinvite=yes in their SIP.CONF 2. Both UAs have same codecs 3. There are no t, T settings in Dial command. I'd like to have a confirmation from * developers about this statement. I.N.
Irakli Natsvlishvili
2005-Sep-07 01:14 UTC
[Asterisk-Users] asterisk, SIP, Re-INVITEs and different contexts
Hello! Hmmm... Folks, I beg you pardon, if I'm telling something which was said before, but actually I have not found this anywhere, neither on Voip-info.org or in several Asterisk's docs. So, here is the statement: If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between them will ALWAYS go via Asterisk. I.e. Asterisk WILL NOT issue Re-INVITE even if: 1. Both UAs have canreinvite=yes in their SIP.CONF 2. Both UAs have same codecs 3. There are no t, T settings in Dial command. I'd like to have a confirmation from * developers about this statement. I.N.
Olle E. Johansson
2005-Sep-07 01:18 UTC
[Asterisk-Users] asterisk, SIP, Re-INVITEs and different contexts
Irakli Natsvlishvili wrote:> If SIP extensions are in DIFFERENT CONTEXTS, then RTP traffic between > them will ALWAYS go via Asterisk.Dial plan contexts has nothing to do with how we set up RTP traffic.> I.e. Asterisk WILL NOT issue Re-INVITE even if: > > 1. Both UAs have canreinvite=yes in their SIP.CONFIf canreinvite=yes, we *will* issue a re-invite if possible.> 2. Both UAs have same codecs > 3. There are no t, T settings in Dial command.Or h,H or nat=yes. It is easier to turn it around: Asterisk will issue a re-invite unless there is a reason to keep the audio going through Asterisk * NAT traversal issues * Canreinvite=no * Anything that needs asterisk to listen for DTMF in call * Codecs that needs to be transcoded /Olle --- Astricon 2005 - where you will learn about Asterisk and re-invites! http://www.astricon.net/2005/ October 12-14 Anaheim, California