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

[Sammy Govind]

I think the time when you were having full audio w/o any RTP proxy was due
to the fact that you've everything on the same subnet.
In my opinion you'll need  RTPproxy eventually whenever you deploy it in
real environments.

I cant tell the exact reason for destination unreachable, maybe the virtual
IP has something to do with it.

To configure RTP proxy you need to do following.
1- Start RTPproxy in bridged mode i.e #*/usr/sbin/rtpproxy -l
<externalipofproxy>/<internalipofproxy> blah blah switches*
2- Set module params in opensips.cfg file, and find out the point in main
route where call is forwarded to Asterisk Server VIP, just before that
 write the function "*force_rtp_proxy("ei")*"
3- hmmm..on BYE or CANCEL you need to unforce rtp proxy as well.

Thats all I could think about it so far.

Regards,
Sammy

On Mon, Dec 19, 2011 at 10:36 PM, Nick Khamis <symack at gmail.com> wrote:

> 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.
>
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