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Sat 17 of May, 2008 [05:28 UTC]

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  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
Server Stats
  • Execution time: 0.37s
  • Memory usage: 2.18MB
  • Database queries: 33
  • GZIP: Disabled
  • Server load: 0.41

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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