[OpenSIPS-Users] Totally Stunned about this No Audio Going Out

[Nick Khamis]

Hello Sammy,

Thank you for your response. I now have outgoing audio again which is
half the battle.
The second half (incoming audio), has proven to be a challenge. Maybe
if I start with
a description of the setup:

* This is a test environment done on virtual machines


Network:

RouterL (192.168.2.1)
Polycom Phone (192.168.2.11)
OpenSIPS (192.168.2.102)
Asterisk Virtual IP for AST1 and AST2 (192.168.2.6)
Asterisk1 (192.168.2.110)
Asterisk2 (192.168.2.111)


-------------  Port FWD (1)    --------------------------------
      ---------------------------
| Router |----------------------> |OpenSIPS/RTPProxy|----------> |
Asterisk GTWY  | ----------- Internet/ITSP
-------------
---------------------------------
---------------------------


1) The port forwarding range is:
     SIP: 5060
     RTP: 10,000-50,000
     RTP Proxy:  7789


I just want to clear some things up. I had outgoing audio the whole
time without RTPProxy.
All the test UC (Polycom Phones) are within the same network. Do I
need to use RTPProxy
to get incomming audio working? As you can see in the diagram, I did
try using RTP Proxy
but never succeeded.

Doing a raw UDP trace from ports (10000-50000) I found this:
http://pastebin.com/yzgBZQ9S
There is a "Destination unreachable" at first attempt being returned
by opensips server,
and then it dissapears, the it comes back again. Not sure if this is
related to the no
outgoing audio, but I will need to resolve it nevertheless.

As for a SIP trace without RTP Proxy proxy running:
http://pastebin.com/PUXJ3wpK.
Wanted to turn your attention to:

* The network architecture consists of OpenSIPS sending requests to
the Asterisk virtual IP (192.168.2.6),
which is connected to the Asterisk physical machines (192.168.2.110,
192.168.2.111). The responding
asterisk box, in this particular eaxample, was 192.168.2.111. I hope
this would not be the problem?

* A summary of the SDP trace is as follows:

INVITE from UC:                       m=audio 10006 RTP/AVP 0 8 18 101
OK from Asterisk to OpenSIPS: m=audio 31576 RTP/AVP 8 0 101.
OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
OK From OpenSIPS to UC:       m=audio 34030 RTP/AVP 8 0 101.
OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.
OK from Asterisk to OpenSIPS: m=audio 34030 RTP/AVP 8 0 101.

Is taht my problem right there? My system is unable to connect
the initial request from the UC on port 10006, to the followup response
of the ITSP on port 34030? There is no OK from the UC to OpenSIPS.

I've been struggling with this for a week now. Any help would be greatly
appreciated!

Kind Regards,

Nick.