For a SIP proxy, the preferred method is indeed to add a Record-Route
In the script, add record_route(); instruction.
Emmanuel
IVèS
Le 2017-03-02 à 20:03, Robert Johnson a écrit :
I seem to recall reading something stating that you shouldn't modify
the contact header - But, I can't find the text.
Might suggest adding a Record-Route header instead of re-writing the
contact header?
I'm curious as to what the list has to say about this question, I've
considered doing the same thing.
On 03/02/2017 04:53 PM, Jack Davis wrote:
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@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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_______________________________________________
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