[Asterisk-Dev] How are SIP calls connected/bridged ?

[Arnaud]

Please help me understand the magic by which Asterisk knows that SIP
Callid1 is "bridged" with SIP Callid2. I've been studiying chan_sip.c
for quite some time and still don't get it. Is the "peer" stored in sip_pvt ?

Scenario:
Assume that initially * bridges SIP Callid 1 with Callid 2 (Callid1 is
between Alice and *, Callid2 is between Bob and *)

Later on * Answers() a SIP call and SIP Callid3 is created. (Callid 3
is between Todd and *)

Under certain circumstances I want * to bridge Callid 1 with Callid3
(that is Callid 3 replaces Callid2).

by briding I mean, route the RTP traffic and the SIP signaling so that
Alice speaks with Todd.

Thanks in advance - Arnaud