[Asterisk-Dev] Change IP address in the middle of a VoIP call

[Arnaud]

We are looking into mobile users that change their IP address in the
middle of a VoIP call. We use SIP and Asterisk does not reinvite the
two ends i.e. Asterisk acts as a B2BUA and sees both the SIP and the
RTP traffic.

We've been searching chan_sip.c, rtp.c and other related files trying
to understand where the following are set/modified
- SIP next hop
- IP/port to use for sending RTP data

A quick pointer on how to change the above in the code would be of
great help. Also are there existing methods to change the above on the
fly?

thanks  - Arnaud