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Record-route will only help for messages that go through
your proxy - if the INVITE is not even hitting your proxy, then you will need
some other mechanism to tell Asterick to send all requests through you. I
don't know asterick at all but maybe you can configure your proxy as
an outbound proxy on asterick? So that it will route all requests
through you first.
Or...I'm not 100% sure if this is possible or not - but you
could try adding an embedded Route header to the Contact Uri of Registrations
that pass through you.
Scott From: Steven Coule [mailto:Steven.Coule@xxxxxxxxx] Sent: Thursday, March 16, 2006 5:25 AM To: Scott Godin; resiprocate-devel@xxxxxxxxxxxxxxxxxxx Subject: RE: [reSIProcate] Routing of signalling via proxy - opinions? I wasn’t intending my
proxy to be the registrar, merely to route the REGISTER requests to the
registrar in addition to the normal proxying behaviour of the call signalling.
As an example scenario,
I want to sit my proxy logically between an Asterisk PBX (registrar) and the SIP
phones it uses so that all calls can be monitored using my
proxy. The tricky bit seems to
be forcing the Asterisk to route calls to the SIP phones via my proxy. It
appears to ignore the maddr parameter and send the INVITEs directly to the
phones. Do you think Record-Route will solve this?
Thanks for the tip
about repro, I’ve downloaded it and am taking a
look.
From: Scott Godin
[mailto:slgodin@xxxxxxxxxxxx] If the UA's register
with your proxy and are only reachable via their AOR - then using Record-Routing
on the proxy is the way to go. You may want to take a
look at repro - it is a proxy built ontop of resiprocate. Repro already
supports record routing - you could just extend repro to add your monitoring
code. Scott From:
resiprocate-devel-bounces@xxxxxxxxxxxxxxxxxxx
[mailto:resiprocate-devel-bounces@xxxxxxxxxxxxxxxxxxx] On Behalf Of Steven Coule I am developing a SIP proxy to
monitor SIP call signalling using reciprocate and need to force all call
signalling traffic via my proxy to maintain an accurate call state
model. For outbound calls from a UA, the UA
can be pointed towards the proxy, so that is straightforward. For inbound calls
to any UA that I need to monitor using my proxy, there needs to be some method
for forcing call signalling to my proxy rather than directly to the UA. For
example, my proxy could sit logically between an Asterisk PBX and a UA for that
PBX, or between a E1->SIP gateway and an array of
UA’s. As far as I understand, there are a
couple of approaches I can use to achieve this .. 1) Use the
maddr= parameter. By forcing the UA to register with my proxy, I can manipulate
the REGISTER by adding the maddr= parameter before forwarding the REGISTER to
the real registrar. In theory at least, this should force the inbound calls to
be routed via my proxy by the Asterisk / SIP
server. 2) Using the
RFC3581 mechanism for NAT traversal appears to provide a method of forcing a
call signalling path via a specific proxy using the Record-Route and rport=
parameter. Have I missed anything? Which method
is preferred?
Steve |