[OpenSIPS-Users] Asterisk reinvite confuses Mediaproxy

[Jeff Pyle]

Hello,

Here's the scenario (no NAT):


              SIP phone
                  |
                  |
   Opensips 1.5.1 & Mediaproxy 2.3.4
                  |
                  |
        Asterisk w/ reinvites
                  |
                  |
             PSTN gateway


engage_media_proxy() happens on all initial INVITEs.

As many of us are too aware, Asterisk starts with relaying the media through
itself.  The canreinvite option allows it to send reinvites to the two
endpoints, adjusting the connection information to allow the endpoints to
send media directly between themselves.  It seems as though these reinvites
are confusing Mediaproxy in some cases.

When an outbound call (SIP phone --> PSTN gateway) happens, all is well.
Mediaproxy reports the caller_remote and callee_remote IP/ports correctly as
the SIP phone and the PSTN gateway, respectively.

An inbound call (PSTN gateway --> SIP phone) is less successful.  After the
reinvite, the SIP phone --> PSTN gateway audio is relayed correctly.  But,
the PSTN gateway --> SIP Phone audio still relays to Asterisk, not to the
SIP phone.  The caller_remote reports as Asterisk's IP/port, not as the PSTN
gateway's.

I see what's happening, but I don't know why.  What's the next logical step
in diagnosing this?


Thanks,
Jeff