Hi Ruud, Sorry for any confusion. I've attached fresh traces, including a full ngrep and mediaproxy relay and dispatcher logs.
This is an inbound call from PSTN gateway to Asterisk (with reinvites) to Opensips with Mediaproxy to the callee endpoint. I have a single engage_media_proxy() at the initial invite. - Jeff On 7/16/09 4:15 AM, "Ruud Klaver" <[email protected]> wrote: > Hi Jeff, > > I've just been scrutinizing your SIP trace, as you still haven't > provided me with mediaproxy-relay debug output. What happens when the > SDP offerer comes with a new ip/port combination for a particular > stream is that mediaproxy allocates a new set of ports for this > internally. You can see that this happens by the fact that for the re- > invite, the RTP port in the modified SDP is different. This means that > both endpoints actually should start sending to a new destination as a > result of the re-INVITE exchange. If they do, the previous RTP > exchange and the next one can never actually "cross wires". > > Now I'm not exactly sure what your problem is, as you said before it's > PSTN -> SIP phone that is giving you trouble, yet you've included a > trace which seems to be in the opposite direction. Again, please > include a media-relay log and describe what you are (not) hearing at > either endpoint. > > Ruud Klaver > AG Projects _______________________________________________ Users mailing list [email protected] http://lists.opensips.org/cgi-bin/mailman/listinfo/users
