[Asterisk-Users] asterisk, SIP, Re-INVITEs and different contexts

[Olle E. Johansson]

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.CONF
If 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

---
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