Benny Amorsen schrieb: > Stefan Schmidt <s...@sil.at> writes: > >> What kind of client cant handle one packet per minute without getting a >> higher load? > > It isn't a client. It handles thousands of connected devices, so it'll > be handling perhaps 50 OPTIONS packets every second if I go the qualify > route.
if i understand you right you have one server (peer) where thousands of devices are connected and every device is registered to asterisk, and so every options packet will come from asterisk to this device, right? If you have a sip routing server like ser, the server itself could do a Nat keep alive check, and could drops the invite coming from asterisk if the peer isnt reachable. If these devices arent registered to asterisk why do you think that there will be so much options Packets? if you have one peer this will get only one Options packet per minute. if you just have an rtp routing server or something similar you should have a look at ser / openser/opensip for handling these devices directly. >> What your are searching for is called Sip T1 Timeout and i´ve seen >> that in asterisk 1.6.1.x you can set this timeout in sip.conf. Iam not >> sure about changing this in other versions. > > If you're talking about t1min, AFAIK that only applies to "monitored > devices", i.e. those with qualify=yes. > > /Benny > i am talking about t1max which is per rfc definition 64xt1min. Which is normally 32000 milliseconds. If you set this down to 15 seconds the timeout would be half than now, but could cause problems with very slow clients. The qualify options only takes affect on t1min when it set to yes. Then t1min would be set to the average qualify value. As i said i think qualify would be the right solution for you. I have a server running with more than 1600 peers, all with qualify on and notify traffic is around 200 pps in the night with no calls and aroung 6kpps (also with rtp traffic) on high load without taking any affect of the system. Our Ser server have a constant load of 600 pps but that is a proxy build for doing nothing else than routing sip packets. best regards steve _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
