[SR-Users] SIP contact header question

[Jack Davis]

Greetings,

I have a general question about the usage of SIP contact headers in the
context of using Kamailio as a SIP proxy.


________________________________

[  A  ] --> [  Kamailio  B ]  ---> [  C ]

Node A originates a SIP invite, containing a valid via header and URI while
setting the contact address to a user at itself and delivers it to Kamailio B
which is acting as a SIP proxy.

Kamailio B then uses dispatcher routing to direct the Invite to node C,
adding a via line with its own information as well as a record-route header
with its own proxy information but retaining the same contact address from
A.

Node C establishes the call and then sends a re-invite to the Kamailio B
proxy which is in turn sent to Node A. Node A responds with a 200 OK

The problem arises when Node C tries to send an Ack in response to this 200
OK. The ack is being sent to the Contact address, rather than the routing
already established in the initial dialog.

________________________________

My question is: should kamailio be rewriting this contact address with its
own? Is that the best practice? My understanding is that the contact header
is more so related to future requests within the same dialog ONLY when a
record-route is not used.

I would appreciate any clarification on the RFC or best practices in this
scenario.

Thank you,
Jack Davis
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