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

[Nick Khamis]

What happened to my nice diagram? Argh.... Sorry guys!

Router -> OpenSIPS -> Asterisk -> ITPS

On Mon, Dec 19, 2011 at 12:20 PM, Nick Khamis <symack at gmail.com> wrote:
> 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.