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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 |