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

[Olle E. Johansson]

SIP calls are not connected to each other. In the Asterisk architecture,
the SIP "call" is connected to an owner channel. Each call has a pair
of a "technology" driver structure (tech_pvt) - like IAX, SIP, H323, ZAP
and they connect to a generic Asterisk channel that is the "owner".

So when Bob calls Alice, both on SIP devices, the incoming SIP call
connects with an Asterisk channel that connects (bridges) to another
Asterisk channel that connects to another outbound SIP call. SIP
signalling always ends in Asterisk and is not forwarded in any way
"through" Asterisk.

So by reading chan_sip, you will not discover much, you need to dig
deeper into the channel and pbx interface. In chan_sip, you can check
the code for transfers that handle all these channels and redirecting
them in various ways.

Good luck exploring!

/Olle