On 24 aug 2010, at 12.16, Tony Graziano wrote: > In looking at the sip trace, I don't see the IP of the tandberg > (192.168.0.116 at all listed).
Frame 24 show the SIP INVITE going from SipXproxy to the Tandberg (behind NAT) address, 85.24.160.19:43272. You will see that the SIP Request URI points to "sip:[email protected], with "X-Sipx-Nat-Route: 85.24.160.19:43272;transport=tcp" that I guess is part of the SipX NAT traversal helper. The Tandberg answers with 180 Ringing (frame 27) and 200 OK (frame 42), delivered to the initiated phone in frame 46 (192.168.0.112 behind NAT, 85.24.160.19:41164 from SipX perspective outside NAT). Now, the initiating SIP UA (Polycom, 192.168.0.112/85.24.160.19:41164) sends the ACK to the GRUU contact specified by the Tandberg E20 in the SIP Contact header (frame 49). The SipXproxy forwards the ACK to SipRegistrar (frame 50) and gets it back in frame 51 with (I guess) the GRUU ID resolved into the SIP URI (sip:[email protected]:43272). This ACK is now "stuck" in SipXProxy (frame 52-70) and never sent out to the Tandberg UA, thus hanging/disconnecting. Since no Route-headers are present in the ACK the Request URI is used for routing and my guess is that something is wrong or un-parsable there since the request isn't forwarded. This is indicated in sipXproxy.log with the "Invalid addr-spec found" attached in previous mail. Or am I missing something else? > I also see a contact at: > > <sip:[email protected];gr;transport=tcp> Yes, this I thought is the registered GRUU ID that the SipRegistrar "resolves"? The Tandberg UA uses (and negotiates) gruu during SIP Registration. > Does the tandberg register somewhere? What is the contact uri for the > tandberg? The Tandberg is registered locally to the same SipXecs server as the Polycom. The registration process is not shown in the trace though but it looks like this in the SipX GUI: sip:[email protected] <sip:[email protected]:43272;transport=tcp;x-sipX-privcontact=192.168.0.116%3A5060%3Btransport%3Dtcp> 97 urn:uuid:736346f2-a91a-5fdd-9102-25cb28ddc2cf Note that the urn might have changed from the traces, but I checked at the time and they were the same as the SipRegistrar replacement. > I also am pretty sure that with sipx you want to run 3.2.3 with sipx on the > 335. 3.2.1 and 3.2.2 have some bugs that result in calls being sent to > voicemail instead of ringing on the 335. Sure thing, but I don't think that's an issue in this case? Can try to update though. /Staffan -- Staffan Kerker mail/sip/xmpp: [email protected] "Don't get involved in politics man, just play the gig..." /Sgt Floyd, Electric Mayhem Band
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