[OpenSIPS-Users] RE-INVITEs being sent to original contact doesn't properly adjust RTP ports on transfer?

[Bogdan-Andrei Iancu]

Hi Tyler,

So ngrep-ing on proxy, you do not see the second re-INVITE (which leads 
to one way audio)....A possibility is that the re-INVITE may by-pass 
your opensips. Do you do record-routing also for sequential requests ? 
There are some bogus UAC/UAS that continuously update the route set, 
even after the dialog was setup. So maybe the first re-InVITE works ok 
as you correctly do RR for initial INVITE, but second re-INVITE fails 
because UAC/UAS expect RR on first re-INVITE too....

Just a supposition

Regards,
Bogdan

Tyler Merritt wrote:
> Hi,
>
> We've got three parties for this example:  A, B, C
>
> A - Asterisk end-point Polycom
>
> B - Asterisk end-point Polycom
>
> C - Outside end-point Uniden
>
> OpenSIPs sits in front of the Asterisk servers and communicates with a 
> carrier C5 switch directly (same local area network inside a lab facility)
>
> A calls C - call completes - talk, no issues.
>
> C presses the transfer button, which is a flash-hook putting A on 
> hold.  C dials B.
>
> B answers the call - both parties talk.
>
> C presses the flash-hook button again in order to complete the transfer.
>
> A can hear B - B cannot hear A.
>
> The RTP debug from Asterisk shows that RTP packets from B are still 
> going to C.
>
> B didn't get the RE-INVITE apparently - but I cannot figure out where 
> the packet is.  It's not showing up in OpenSIPs sip_trace, and it's 
> definitely not getting to Asterisk.
>
> I don't have control of the Carrier-side C5 to check, and they have 
> been slow to provide me with a wireshark trace.  
>
> Is there anything else I could do in OpenSIPs to determine if the 
> RE-INVITE is not being handled properly besides what I've already 
> mentioned?
>
> Thanks in advance.
>
> Tyler

-- 
Bogdan-Andrei Iancu
OpenSIPS Event - expo, conf, social, bootcamp
2 - 4 February 2011, ITExpo, Miami,  USA
OpenSIPS solutions and "know-how"