What happened to my nice diagram? Argh.... Sorry guys! Router -> OpenSIPS -> Asterisk -> ITPS
On Mon, Dec 19, 2011 at 12:20 PM, Nick Khamis <sym...@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. _______________________________________________ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
