If you're having to think about how to do things that break basic SIP semantics, it
may be time to rethink your design.
More particularly, passing requests from A to B back to A, when A and B are both proxies,
is problematic. It will lead to potential call loops if the request should find itself
back at the same UA that originated the call. The usual solution is to create a B-leg by
laundering the call through a lightweight, signalling-only B2BUA, such as SEMS.
-- Alex
On 27 Nov 2023, at 08:04, Benoit Panizzon via sr-users
<sr-users(a)lists.kamailio.org> wrote:
Hi List
Two Kamailio Nodes situation.
Node A: Routing Instance.
Node B: Registrar Instance.
An invite is sent from Node A to B.
Customer registered on B is 'busy' as example.
B initiates Call Forwarding by adding a Diversion Header and sending
the Invite back to A with a new R-URI towards the CFB destination.
The Invite sent from B to A still has Node B in it's Record-Route. So
all subsequent SIP messages pass via B, which is not needed and cause
issue with rtpengine running on B.
I attempted to remove_record_route(); when sending the call back to A,
but this causes:
parse_headers(): bad header field [;lr;ftag=3910078620-883101371>
It looks like only the beginning of the RR header is removed leaving
back a competely broken line.
What is the propper was to remove a node from RR and Via routing?
Mit freundlichen Grüssen
-Benoît Panizzon-
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