Sorry, some extra info:
10.1.248.31 is the GW (patton).
The Patton doesn't like to use the same port (5060) to 2 different SipX
clusters.
So I configured it to use port 5061 as source port for my second SipX
HA-pair.
See part of the Patton config below:
context sip-gateway gssipx02
interface TH-sip
bind interface LAN context router port 5060
context sip-gateway gssipx02
no shutdown
context sip-gateway mssipx02
interface MN-sip
bind interface LAN context router port 5061
10.31.48.25 is server A and 10.32.48.25 is server B
10.3.203.150 is the PC with Bria.
I think things start to go wrong in packet 23, that Syn should have gone
to port 30300 (the UDP port used by Bria to server A, port 5060, see
capture A)
and not to 1312 (the TCP port used by Bria towards server A, port 5060,
again, see capture A).
Paul
From:
Tony Graziano <[email protected]>
To:
Discussion list for users of sipXecs software
<[email protected]>
Date:
07-02-2011 11:59
Subject:
Re: [sipx-users] Problems with HA-setup, UDP and TCP ports mixed up, only
50% of calls come through
Sent by:
[email protected]
its opening a tls connection (port 5061)?
frame 21 (capture b) shows the source port for the invite to come from
sipx on port 5061, and i (also) don't understand why it would do that.
On Mon, Feb 7, 2011 at 5:34 AM, <[email protected]> wrote:
Maybe someone can look at the traces attached.
The trace ServerA.pcap is taken on the sipserver that holds the
registration. This server is sending the invite over the existing session
and all works.
The trace ServerB.pcap is taken on the other server and this one tries to
open a session to the phone on the wrong port (if I am not mistaken).
This is RST-ed in packet 24 and then S-B sends an invite to the IVR.
If wanted I can send a snapshot from S-B or both servers.
Please advise what to do.
Paul
From:
[email protected]
To:
[email protected]
Date:
04-02-2011 16:07
Subject:
[sipx-users] Problems with HA-setup, UDP and TCP ports mixed up, only 50%
of calls come through
Sent by:
[email protected]
For the second time in a looong time I have the problem that only 50% of
the calls from a GW to a SIP-phone come through, the other 50% go to
voicemail directly.
I have an HA setup and the GW is distributing the calls round-robin to the
2 SipX servers that form an HA-cluster.
Also calls from phones registered on one server (S-A) of the HA cluster to
a phone with the problem on the other server (S-B) will go to voice mail
directly.
I am using 4.2.1-018971 and Bria 3.1.2.1
Most phones (Bria's) work OK, but some show the following behaviour:
When the call flows to the SipX server (S-A) that holds the registration
for that phone then the phone starts ringing.
When the call flows to the other SipX server (S-B) then you go to
voicemail directly.
I have traced a working Bria and one with the problem. The difference lies
in the port that is being used to send the invite on.
BTW: my phones are configured to use TCP.
In the normal case the phone registers from port A to 5060 (syn, syn ack
etc) on S-A.
The phone then sends some Notify requests over UDP from port B to 5060 on
S-A.
If the call flows through S-A (the server that holds the registration)
then the existing session (5060 to port A) is used.
If the call flows through S-B then the server sends a SYN from a random
port to the phone to port B (so the same port that the phone used to
communicate with S-A for UDP communication)
This is SYN-acked and all goes well.
In the case of the problem all is the same for S-A and these 50% of the
calls work.
If the call flows through S-B however then the SYN is not send to port B
(the UDP port used by the phone between phone and S-A), but to port A (the
TCP port of S-A).
This SYN is generously RST-ed and you end up in voicemail.
I found a really old issue, http://track.sipfoundry.org/browse/XCL-89
maybe something like this can happen very rarely as well.
If needed I can send wiresharks, snapshots or anything else, please let me
know.
Restarting services does not help.
Resetting the servers one at a time does not solve the problem as well.
The only remedy so far that sort of works is resetting both servers at the
same time.
Paul_______________________________________________
sipx-users mailing list
[email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-users/
_______________________________________________
sipx-users mailing list
[email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-users/
--
======================
Tony Graziano, Manager
Telephone: 434.984.8430
sip: [email protected]
Fax: 434.326.5325
Email: [email protected]
LAN/Telephony/Security and Control Systems Helpdesk:
Telephone: 434.984.8426
sip: [email protected]
Helpdesk Contract Customers:
http://support.myitdepartment.net
Blog:
http://blog.myitdepartment.net
Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4
_______________________________________________
sipx-users mailing list
[email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-users/
_______________________________________________
sipx-users mailing list
[email protected]
List Archive: http://list.sipfoundry.org/archive/sipx-users/
