>> 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]
<mailto:[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
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