Re: [SR-Users] need help for newbie in Kamailio

-- Inutile d'imprimer ce mail 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@gmail.com > mailto:reda.aouad@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@gmail.com > mailto:govoiper@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@gmail.com mailto:mvineetmenon@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@gulfsat.mg > mailto:teddy@gulfsat.mg> wrote: > > Hi mailing, > > Newbie to kamailio, I follow this tuto > http://nil.uniza.sk/sip/kamailio/adding-mysql-support-kamailio-31-debian-len... > 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. > > -- > Inutile d'imprimer ce mail > > > _______________________________________________ >