[asterisk-users] Reinvites and SIP/RTP

[Martin Schuhmacher]

Noah Miller schrieb:
> Hi Adrian -
> 
>> When I use re-invite, does the Asterisk server stay in the SIP conversation,
>> and just RTP traffic diverts, or does the SIP transfer away from the A*k
>> server too ?
> 
> I'm sure somebody will correct me if this is wrong, but I believe the
> signalling must stay with asterisk, as asterisk needs to know if it
> should provide any services for the call (music on hold, transfer,
> etc).

yes, 'only' rtp goes direct, SIP stay on asterisk since it might
be a hangup or something else comes in.

Yours
Martin