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, <pscheep...@epo.org> 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: pscheep...@epo.org To: sipx-users@list.sipfoundry.org 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:
> sipx-users-boun...@list.sipfoundry.org
> ------------------------------
>
>
>
> 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*<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_______________________________________________
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>
> _______________________________________________
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> sipx-users@list.sipfoundry.org
> List Archive: http://list.sipfoundry.org/archive/sipx-users/
>



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