[SR-Users] need help for newbie in Kamailio

Rabary teddy at gulfsat.mg
Wed Mar 21 07:50:45 CET 2012


Thank you all for the answers. Finally, there is not a very big problem 
if one side is disconnected because the other side will end dialog when 
phone will be hang up

have a good day :)

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Le -10/01/-28163 22:59, Reda Aouad a écrit :
> Not also that the recommend session timer value that UACs generally 
> use by default is 1800 seconds, or 30 minutes. Just in case you may 
> accept it. To me, it's not acceptable when customers are billed.
>
> However, the risk is not so big. If one of the UACs is suddenly 
> disconnected, the other party will most likely hang up, the dialog is 
> terminated and finally we loose only a few seconds of billing.
>
> Reda
>
>
>
> On Tue, Mar 20, 2012 at 10:02, Reda Aouad <reda.aouad at gmail.com 
> <mailto:reda.aouad at gmail.com>> wrote:
>
>     Hello,
>
>     In SIP, session timers can be used to periodically ping the UAC
>     (using a re-INVITE or UPDATE) to know if it's alive or not. Then
>     action can be taken - terminating the call.
>
>     Kamailio has the SST (SIP Session Timer) module which only
>     enforces a minimum session timer value for UACs, but not a maximum
>     one. It doesn't ping the UACs neither. This is fine because the
>     RFC stops here. A nice improvement to Kamailio would be to augment
>     the SST module with a feature which enforces a maximum session
>     timer value and pings UACs. Another suggestion would be to rely on
>     nathelper's keepalive results to take a decision after a keepalive
>     times out, but then we'd have to terminate all dialogs in which
>     the UAC that is not responding is present, since nathelper's
>     keepalive are out-of-dialog. No very neat, but functional.
>
>     And I don't think the dialog module can do anything about this
>     problem.
>
>     I know that what I am suggesting may not be defined in RFCs, and
>     so are some features of SIP servers, but in my opinion should be
>     implemented as it adds a great value to Kamailio.
>
>     We cannot rely on RTP timeout since a UAC may use a
>     silence-detection codec and be silent for some time, or may put a
>     call on hold for a while, not sending RTP packets in both cases.
>     This is why RTP timeout detection is not reliable. Anyway,
>     mediaproxy timeouts ONLY AND ONLY in the case it doesn't receive
>     RTP packets from BOTH UACs, not only one, for the reasons
>     mentioned. I don't know about rtppoxy, maybe others can tell more
>     about it.
>
>     One solution if you really need to solve your problem would be to
>     put a B2BUA in the SIP path, such as Asterisk or FreeSwitch. They
>     enforce a maximum session timer which UACs can use to ping
>     themselves every now and then, and Asterisk can even ping the UACs
>     and terminate the call if one of them doesn't respond. The
>     downside: lower performance and higher cost. Asterisk is very
>     heavy and Kamailio can handle many, many more calls, so you'll
>     have to load balance to several Asterisk servers if you have a
>     single Kamailio machine handling thousands of simultaneous calls.
>
>     Kamailio developers out there, what about boosting the SST module
>     with new features? Or creating an SSTX module?
>
>     Reda
>
>
>
>     On Tue, Mar 20, 2012 at 07:35, SamyGo <govoiper at gmail.com
>     <mailto:govoiper at gmail.com>> wrote:
>
>         Hi,
>
>         Yes that is the behaviour when the media isn't flowing through
>         a regulatory tool (in-terms it sees the media and know call is
>         actually going on rtpproxy/media-proxy)  but in the absence of
>         any such tool SIP server is not aware that the call-media is
>         still in progress or is dead ! so it always assume that the
>         call is active and hence the BYE signals are never originated
>         from server end to shutdown the call.
>
>         I am definitely not an expert but I am guessing
>         that dialogue module do some keepalive tests for an ongoing
>         session and not sure what it do if either end fails to respond !!
>
>         Regards,
>         Sammy
>
>         On Tue, Mar 20, 2012 at 11:04 AM, Vineet Menon
>         <mvineetmenon at gmail.com <mailto:mvineetmenon at gmail.com>> wrote:
>
>             i guess it should time out...the other end...since it has
>             no way of knowing that the other end is no more present...
>
>             Regards,
>
>             Vineet Menon
>
>
>
>
>
>             On 20 March 2012 11:30, Rabary <teddy at gulfsat.mg
>             <mailto:teddy at gulfsat.mg>> wrote:
>
>                 Hi mailing,
>
>                 Newbie to kamailio, I follow this tuto
>                 http://nil.uniza.sk/sip/kamailio/adding-mysql-support-kamailio-31-debian-lenny
>                 for the registration SIP via mysql database and it
>                 works fine, but I saw that when during the call we
>                 disconnect the called UAC from network or turn the
>                 power off the caller UAC don't hangup.
>                 Is there any tool for how to hangup call when the UAC
>                 on the other side has no network connection or it
>                 isn't power on durring a call ?
>                 I heard for mediaroxy or rtpproxy but I don't know if
>                 them can do what I except to haveand we also use ip
>                 routing to make kamailio server to communicate with
>                 the UAC so we don't use NAT.
>
>                 Our topology is:
>                 kamailio (with public IP address) ---> cisco switch
>                 ---> LAN ---> UAC (with private IP address)
>
>                 Thanks in advance.
>
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>
>
>                 _______________________________________________
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