[SR-Users] SIP contact header question
robert.j at bendtel.com
Fri Mar 3 02:03:51 CET 2017
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
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:
> 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
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users at lists.sip-router.org
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