Hello,

I'm testing freeswitch as an sbc (version 1.2.5.3). I think I discovered a bug, but I still have to verify it against git if I can find the time to compile it. Anyway, it all relates to session-timer according to me, but even if I disable session-timers, freeswitch still sends sip-update-messages to the sip-client. Freeswitch is dropping the calls after 30 seconds because of this. There is no problem with natting, firewalling, ... as far as I can tell as I'm tracing the packets on all ends. A packet from freeswitch to the sip-client (phone behind other B2B-server):

 UPDATE sip:10000@lab2.ddm.priv SIP/2.0
   Via: SIP/2.0/UDP 172.16.105.14;rport;branch=
z9hG4bKS2pQKQ7SrNgce
   Max-Forwards: 70
   From: <sip:8888@172.16.105.17:5060>;tag=X5FKtKj9ZHaDc
   To: <sip:10000@lab2.ddm.priv>;tag=849E1A5B-6750-4EDE-B7CF-72DF979FD126-40026
   Call-ID: 5AC29923-74EE-46B2-830A-9CB78B9DB47F-39541@172.16.105.17
   CSeq: 38492344 UPDATE
   Contact: <sip:8888@172.16.105.14:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.2.5.3+git~20121206T050429Z~91eef34d5c
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY
   Supported: precondition, path, replaces
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 249
   P-Asserted-Identity: "Outbound Call" <0001>

   v=0
   o=FreeSWITCH 1357636566 1357636567 IN IP4 172.16.105.14
   s=FreeSWITCH
   c=IN IP4 172.16.105.14
   t=0 0
   m=audio 20250 RTP/AVP 9 127
   a=rtpmap:9 G722/8000
   a=rtpmap:127 telephone-event/8000
   a=fmtp:127 0-16
   a=silenceSupp:off - - - -
   a=ptime:20



The other end of the sbc/client doesn't support session-timers or update (yate client), and it doesn't have this problem.

So, the main question is:

--> Why is freeswitch sending these update-messages while the codec, ... remains the same as far as I can tell. It isn't a session-timer, because it is enabled in the profile and it doesn't appear in the message too (which means the disable had some influence ...)?

So, is there a way to:
 1) fix the above?
 2) force session-timers to use reinvite instead of update. As this case may be a way around the bug (freeswitch doing as it is the refresher while they agreed that the client is the session-timer-refresher).
 3) (asked before, but not answered) Is there a way to disable sip-"update"-message-support in freeswitch?

Thank you and kind regards,

Michel