«

Hi,

I'm not sure if this is the best ng to be asking this in but...

I'm trying to understand how routing works in SIP, and in particular how
dynamic networks are handled. I understand that in order to route a call
using SIP is done via a next hop basis similar to IP, but what happens when
the network is changed, how is the routing information in the SIP proxy
updated?

I'd also be interested in is there any provision in SIP for retransmission
of messages as I have done a few tests and it seems that if a call setup
message is lost then a call will timeout. I'm not sure if this was a
configuration issue or just how SIP works as I'm not very familiar with it.
The network setup I used had X-Lite as the SIP client and party-sip as the
SIP proxy. I should have some Cisco 7960 phones to play with soon though.

I've tried looking at the RFCs but there does seem to be an awful lot to
wade through, so I was hoping there maybe some other good sources of
information of where to start looking.

Many thanks

Pete Calvert

--
Using M2, Opera's revolutionary e-mail client: http://www.opera.com/m2/