Re: [SR-Users] dialog - keep alives with lower cseq

Hello, several days ago, just before freezing the development for v3.3.0, I added to dialog module the option to send keep alives for ongoing calls in order to detect if caller/callee is gone. SIP specs require to increment the CSeq for requests within dialog, but because the keep alives are sent from the sip server, it would results de-synchronization of the CSeq values hold in phones themselves (e.g., a BYE created by caller/callee after a keep alive will be with lower cseq than the other side would expect and accept). One solution would be to update cseq when BYE passes the server to the right value. This implies more processing, as a call can have many re-INVITEs or other requests within dialog sent by caller/callee, including periodic updates to database to store cseq numbers. So I went for a different approach, like stated in subject -- let's see your opinion if you think is going to work. Practically the keep alives will be OPTIONS with CSeq equal or lower than the last valid value (e.g., the cseq of the INVITE creating the call). If the caller/callee is gone, that is simply, the OPTION will be timed out and dialog module will send BYE. If the caller/callee are reachable, but for some reason the call was destroyed (e.g., a crash and restart meanwhile), since there is a To-tag, the OPTIONS should get a 481 Call/Leg transaction does not exists. Again, a case when kamailio will end the dialog. The one to be discussed here would be caller/callee are still on the call. Based on RFC (and the feedback from people on another thread here), a requests coming within a dialog with lower CSEq should be replies with 500. CSeq numbers are still ok in both sides (note that requests with lower CSeq can happen in reality, like two fast re-INVITEs sent over UDP, the second arriving first due to network transmission). I tested with snom phones and jitsi so far, seems to be a working solution (well, jitsi was replying 500 after 20secs to keep alive OPTIONS request, not sure for what reason, just reported back to the project). Is anyone here seeing any possible issues with the approach? Call are ended by dialog, with proper CSeq in the BYE, after 10seconds from the moment 408 or 481 is received, no other replies are taken in consideration for ending the dialog from server side. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users