SIP CANCEL not working with Win32 RTC based SIP phones

[ittiam]

1. Brekeke Product Name and version:
Brekeke SIP Server (bundled with PBX 2.0.7.2)

2. Java version:
1.5

3. OS type and the version:
Windows XP Professional

4. UA (phone), gateway or other hardware/software involved:
Windows RTC based clients

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/networkpatterns.html :
Pattern 1

6. Your problem:
Two Windows RTC based SIP phones PhoneA and PhoneB are registered with Brekeke SIP server. PhoneA makes a call (SIP INVITE) to PhoneB, but decides to cancel (SIP CANCEL) the call before PhoneB picks up. In this case, PhoneB does not stop ringing. i.e., PhoneB is not able to understand the CANCEL message relayed via Brekeke SIP.

We have captured the SIP packets at the servers network interface. We see that the CANCEL that originates from PhoneA is being relayed to PhoneB. When PhoneB receives this, it returns a "481 Call Leg/Transaction Does Not Exist" error message, and the call continues to ring (since the dialog has not been cancelled).

This problem does not occur when the call is made directly between the two phones. The CANCEL works properly.

Has anyone come across such a problem before? Can you let me know if there is some way to solve this. Any help appreciated

Notes:
- This installation of Brekeke PBX has been upgraded from a 1.5.X to a 2.0.X installation as per the upgrade instructions on your website.

- We have also noticed that the CSeq in the SIP header of the CANCEL message is being incremented by Brekeke SIP server. Could this be a source of the problem?

[Cross posting this from http://www.brekeke-sip.com/bbs/viewtopic.php?t=5168 in case someone who visits this forum more often has an answer]

[peng]

Please try latest 2.x beta and let us know how it works.

[ittiam]

Hi peng

Thanks for the suggestion. We will try it immediately and let you know.

In the meantime, we have noticed that the CANCEL succeeds with other SIP Clients. The CSeq of the CANCEL is not being modified when we call with other SIP clients like, say, SJPhone.