Jeff, I think the confusion is that some sip components are on the 192.168.53.x network, while others are on 172.21.201.x network. In your description below, you state "which routes", which is indicating there is a router on the network, and the IP addressing would indicate they are on different networks. A router is required to route from one network to the other network.
NAT would be associated with a router - hence the reason why the questions around where the router is. one Polycom extension 1821 at 172.21.201.39 calls another, 7821, at 172.21.201.60. 7821 does an attended transfer to 2164550549, which routes through the Adtran gateway at 192.168.53.11. The sipX system itself is at 192.168.54.46. From: [email protected] [mailto:[email protected]] On Behalf Of Jeff Pyle Sent: Tuesday, September 25, 2012 10:35 AM To: Joegen Baclor Cc: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Incorrect SDP from sipX after xfer completion Joegen, I fully understand my network so I don't see where the problem is. :) All the details, including an adequate topology description, are now in XX-10464 <http://track.sipfoundry.org/browse/XX-10464> . - Jeff On Tue, Sep 25, 2012 at 12:57 PM, Joegen Baclor <[email protected]> wrote: Jeff, I might be missing something but the SDP you pasted o=- 1348230370 1348230371 IN IP4 172.21.201.60 s=Polycom IP Phone c=IN IP4 172.21.201.60 t=0 0 m=audio 30248 RTP/AVP 9 0 8 18 101 c=IN IP4 192.168.54.46 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=x-sipx-ntap:X192.168.54.46-[PUBLIC-IP];54 172.21.201.60 being the original IP of the polycom and 192.168.54.46 being the IP of sipX I assumed that there is a firewall linking these two networks. Maybe a packet capture and a topology description would save us the need to speculate much :-) On 09/26/2012 12:45 AM, Jeff Pyle wrote: Joegen, In my case all the SIP-speaking components are directly routable to each other with no firewalls and therefore no NAT in between. I don't understand what you mean by "sipX being behind a firewall". Is that relevant to the NAT traversal issue you mentioned? - Jeff On Tue, Sep 25, 2012 at 12:42 PM, Joegen Baclor <[email protected]> wrote: >> In my configuration there is no NAT whatsoever. Is there a way to disable NAT traversal completely, thereby working around this issue for the time being? I should also add sipX being behind a firewall. On 09/26/2012 12:15 AM, Jeff Pyle wrote: Joegen, Yes, exactly. INVITE with no SDP. I'll open a tracker shortly. In my configuration there is no NAT whatsoever. Is there a way to disable NAT traversal completely, thereby working around this issue for the time being? - Jeff On Tue, Sep 25, 2012 at 12:09 PM, Joegen Baclor <[email protected]> wrote: Jeff, good bug reporting! By late negotiation, do you mean INVITE with no SDP? There is a known issue with NAT traversal plugin not being able to handle this properly. If you don't mind, please open a tracker in jira and attach packet captures. On 09/25/2012 11:57 PM, Jeff Pyle wrote: Hello, At the end of another thread <http://list.sipfoundry.org/archive/sipx-users/msg41614.html> Tony suggested I try attended transfers and some other parking related operations against an Adtran TA900-series gateway. It seemed there had been some friction here in the past. I was able to test attended and unattended transfers with sipX 4.6 on an Adtran TA908E. The global config option "voice transfer-mode network" is required to support REFERs. Without it the results will be undesirable. With that line, however, unattended transfers worked just fine Attended transfers yielded no audio when the transfer completed if the Adtran is the C-leg of the transfer. Here's why: This is the SDP of the INVITE with Replaces that arrives at the Adtran from sipX to wrap up the transfer: v=0 o=- 1348230370 1348230370 IN IP4 172.21.201.60 s=Polycom IP Phone c=IN IP4 172.21.201.60 t=0 0 a=sendrecv m=audio 2250 RTP/AVP 9 0 8 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 This is correct, telling the Adtran to send audio to 172.21.201.60 (c= line) on port 2250 (m= line). 172.21.201.60 is the IP address of the Polycom who was the A-leg of the call. As part of some DSP cleanup ops the Adtran starts a reINVITE transaction with late negotiation as soon as the above transaction is completed. The offer SDP on sipX's 200 OK of that transaction looks like this: v=0 o=- 1348230370 1348230371 IN IP4 172.21.201.60 s=Polycom IP Phone c=IN IP4 172.21.201.60 t=0 0 m=audio 30248 RTP/AVP 9 0 8 18 101 c=IN IP4 192.168.54.46 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=x-sipx-ntap:X192.168.54.46-[PUBLIC-IP];54 Notice the c= and m= lines have changed. sipX is now telling the Adtran to send audio to 192.168.54.46, the IP address of the sipX instance itself. This is why the A-leg Polycom at 172.21.201.60 doesn't receive any audio after the transfer - sipX told the Adtran to send the audio elsewhere. Any idea why sipX might do that? As a bandaid, one can prevent the Adtran from sending the reINVITE by adding "no prefer reinvite-without-sdp" to the voice trunk configured to talk to sipX. This command is available only in prerelease code at the moment. I believe it will be GA by the end of the month. - Jeff _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
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