Ok, now I see all signalling messages in pcap files. (unfortunately your xml trace file does not contain the whole call...).
I do not change my opinion: You do not have voice because one of your polycom phones (200) does not send Ack message. Or to be precise: phone 201 sends 200 ok, but does not get Ack, that’s why it (phone 201) does not send rtp stream, and may even not play into it's handset received rtp stream. Regarding "phones talks directly to each other". RTP stream is to be sent directly between the phones. But all signaling (SIP) messages is to be sent through sipx. Sip standard provides special header for that: Record-route. Lets consider your call 201 -> 200. You can hear voice, but I still think that there is something GROSSLY wrong there. Look at the packet 16 - Invite from sipx to 200. It contains 3 via headers. The first one (count from the bottom) is set by the originating phone. The other two are set by sipxproxy process, and must contain sipx address, i.e. 192.0.2.10., but there is 192.0.2.254, which is polycom phone. Moreover there is two record-route headers with 192.0.2.254 address. This headers are set by sipxproxy process too. Looks like sipxproxy process thinks, that it has address 192.0.2.254. Which is GGRROOSSSSLLYY wrong. It may happen if you (by mistake) configured 192.0.2.254 address somewhere in sipx configuration. What is in /etc/sipxpbx/sipXproxy-config file? (Do not edit it manually). Of course the above is just a guess, that I can make looking at your trace... Rgds, Nikolay. > -----Original Message----- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of > Claas Hilbrecht > Sent: Friday, February 04, 2011 1:52 PM > To: Discussion list for users of sipXecs software > Subject: Re: [sipx-users] phone INVITE -> TRYING -> RINGING > -> connect? but no audio/voice with PolyCom SoundPoint 650 > > Hello Nikolay, > > thanks for the response. > > > I would guess that you do not have voice when calling > 200->201 because > > there is no Ack sent by "200-phone". And in the trace there > is one way > > RTP. I would say that you should have "one way audio", not > "no audio". > > I > > I omitted RTP in the trace at the first post to save space > but here is the full log. > > > don't use polycom phones and I can't advice if you use appropriate > > firmware version. > > The IRC (#sipxecs) folks told me to use this version. > > > I see another strange thing in the trace. Sipx always sets > > Record-route header, pointing to itself, so that all messages go > > through sipx. I do > > Hmm, maybe this is because of HD audio? Reading > <http://wiki.sipfoundry.org/display/sipXecs/Polycom#Polycom-HD Voice> it seems that the PolyCom phones talks directly to each > other. > > > not see this header in the trace you provided, and I see > Bye message > > sent directly from one phone to another. That is bad. There is > > something wrong in your setup. Search the wiki for topology > > description and setup examples... > > I think my setup is the most simple one. I have LAN > 192.0.2.0/24 dedicated to VoiP. DNS and DHCP are handeled by > sipXecs. A SmartNode 4638 is used to connect sipXecs to our > PSTN lines. Only VoIP devices are connected to the lan. > > > Your traces does not show full message flow. > > Did you take traces from the phones (mirrored ports)? > > For debugging cases like this one I use a plain old 100 MBit/ > HUB (yes a HUB, no switch). So all traffic is "mirroed" to > all ports. Makes debugging with tcpdump/wireshark much more > easier. But as I said before I use WireShark to filter only the call. > > > I'd better just run "tcpdump -s 0 -w filename.cap" at the > sixp server > > and then transfer filename.cap to your pc and view it with > wireshark. > > You'll see full message flow... Phone1 <-> sipx <-> phone2. > (Of course > > you'll see only packets that go through sipx). Or you may > want to use > > sipviewer > > > http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+us > > ing > > +Sipviewer It will show interprocess sipx communication, > which is very > > useful for troubleshooting. > > Attached you will find a "no-audio.xml" for the sipXviewer. > > Thanks for your help > > Mit freundlichem Gruss > Claas Hilbrecht > > -- > http://www.linum.com mailto: claas.hilbre...@linum.com > Linum Software GmbH Langer Wall 5, 37574 Einbeck, Germany > Tel: +49-5561-926730 Fax: +49-5561-926750 Handelsregister > Amtsgericht Göttingen HRB 131128 Geschäftsführer Claas-Jörg Hilbrecht > _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
