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]
<mailto:[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] <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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