[OpenSIPS-Users] mediaproxy/rtpproxy problem with reinvites

Jeff Pyle jpyle at fidelityvoice.com
Mon Nov 3 21:16:58 CET 2008

Makes sense to me.  I have several to choose from.  :)

In the in-dialog section, if I replace your force_rtp_proxy("l") with an unforce-then-force (no flags), I can restore the on-net operation.  It seems to work 100% of the time.  However, on calls from my PSTN gateway to the NATed user (via Asterisk), it works about 20% of the time.

In every case the SDPs for the invites and reinvites look appropriate.  As such, all the gateways and UAs engaged in the call send RTP to the proper place (to rtpproxy).  The problem seems to be where rtpproxy relays the RTP, if anywhere at all, after the reinvites take place.

- Jeff


-----Original Message-----
From: users-bounces at lists.opensips.org [mailto:users-bounces at lists.opensips.org] On Behalf Of Iñaki Baz Castillo
Sent: Monday, November 03, 2008 2:57 PM
To: users at lists.opensips.org
Subject: Re: [OpenSIPS-Users] mediaproxy/rtpproxy problem with reinvites

El Lunes, 3 de Noviembre de 2008, Jeff Pyle escribió:
> I've inserted a bunch of xlogs to see what's happening.

At this point, the most efective is to capture a SIP trace and examine it.

Iñaki Baz Castillo

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