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  • Aykut, Wed 01 of Aug, 2007 [07:53 UTC]: Hi all, does anybody know about Thomson ST2030 SIP phone. I have upgraded it to latest version (1.56) but "Hold" and "Conf" features are not working after the upgrade ?? Do you know any solution or do you have Ver. 1.52 ?? Where can I find it?
  • Edward J Brown, Tue 31 of Jul, 2007 [23:33 UTC]: Has anybody experienced Choppy voice quality when using a Linksys SPA942 in an Asterisk Conference bridge? It works fine with my polycom and Cisco, but sucks with my Linksys.
  • www.astawerks.com, Fri 27 of Jul, 2007 [18:00 UTC]: does anyone use asterisk on top of clark connect? does it work good?
  • simon, Fri 27 of Jul, 2007 [14:16 UTC]: Hi All, Has anyone here managed to get the Cisco79x1 to successfully fail over to the backup proxy. I have 2 asterisk servers , handsets all register and function, except that backup proxy function doesn't work. Any working example would be very apprecia
  • Matthew Richmond, Thu 26 of Jul, 2007 [03:40 UTC]: using the page() application to page across our building...often the meetme conferences don't disconnect after the caller hangs up. Anyone else having this problem. (using Polycom phones)
  • Matthew Richmond, Wed 25 of Jul, 2007 [02:58 UTC]: thanks Nicholas Blasgen! I haven't worked with AGI before, but there's always a first! Thanks again!
  • Nicholas Blasgen, Tue 24 of Jul, 2007 [19:18 UTC]: Matthew Richmond, AGI will handle all that for you.
  • sam, Mon 23 of Jul, 2007 [16:39 UTC]: need help - certain voicemail extension will stop working and recording voicemail on asterisk - anyone know why and how to fix it? Thanks
  • john haji, Mon 23 of Jul, 2007 [14:55 UTC]: free calls to pakistan
  • bong, Sat 21 of Jul, 2007 [19:09 UTC]: hi good day to all can anyone help me how to configured the nortel sip to the signaling server and how to activate in mobile w/ sip compatible without mcs
Server Stats
  • Execution time: 0.25s
  • Memory usage: 2.18MB
  • Database queries: 29
  • GZIP: Disabled
  • Server load: 2.74

SER module sl

The SL module allows ser to act as a stateless UA server and generate replies to SIP requests without keeping state. That is beneficial in many scenarios, in which you wish not to burden server's memory and scale well.
  
The SL module needs to filter ACKs sent after a local stateless reply to an INVITE was generated. To recognize such ACKs, ser adds a special "signature" in to-tags. This signature is sought for in incoming ACKs, and if included, the ACKs are absorbed.

To speed up the filtering process, the module uses a timeout mechanism. When a reply is sent, a timer us set. As time as the timer is valid, The incoming ACK requests will be checked using TO tag value Once the timer expires, all the ACK are let through - a long time passed till it sent a reply, so it does not expect any ACK that have to be blocked.

The ACK filtering may fail in some rare cases. If you think these matter to you, better use stateful processing ( tm module ) for INVITE processing. Particularly, the problem happens when a UA sends an INVITE which already has a to-tag in it (e.g., a re-INVITE ) and SER want to reply to it. Than, it will keep the current to-tag, which will be mirrored in ACK. SER will not see its signature and forward the ACK downstream. Caused harm is not bad--just a useless ACK is forwarded.


Back to SIP Express Router
Created by oej, Last modification by oej on Fri 12 of Sep, 2003 [19:29 UTC]

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