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Wed 01 of Aug, 2007 [09:24 UTC]

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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: 0.35s
  • Memory usage: 2.18MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 2.78

SER load balancing

From the SER administrator's manual



The SIP standard's use of DNS SRV records has been explicitly constructed to handle with server failures. There may be multiple servers responsible for a domain and referred to by DNS. If it is impossible to communicate with a primary server, a client can proceed to another one. Backup servers may be located in a different geographic area to minimize risk caused by areal operational disasters: lack of power, flooding, earthquake, etc.

Unfortunately, at the moment of writing this documentation (end of December 2002) only very few SIP products actually implement the DNS fail-over mechanism. Unless networks with SIP devices supporting this mechanism are built, alternative mechanisms must be used to force clients to use backup servers. Such a mechanism is disconnecting primary server and replacing it with a backup server locally. It unfortunately precludes geographic dispersion and requires network multihoming to avoid dependency on single IP access. Another method is to update DNS when failure of the primary server is detected. The primary drawback of this method is its latency: it may take long time until all clients learn to use the new server.


This means that
  • You can set SRV records of a domain to point to a SIP proxy in another domain. Like e-mail, where a mail server handles mail to many domains, a SIP proxy can handle users in many domains. The key is to set the DNS SRV records in the user domain (like voip-info.org) zone file to point to a SIP Proxy somewhere else (like sip.iptel.org). With a configuration like this, calls to user@voip-info.org will be automatically routed to the SIP proxy sip.iptel.org.
  • In a DNS zone, you can specify multiple SIP proxies. If the proxy with the highest priority is not reachable, the SIP ua or proxy that tries to reach a user within the domain tries the next proxy defined in the zone with a SRV record.



No one really mails user@mailserver any more. We're mailing user@domain and the DNS MX records helps the mail client to send the mail to the correct mail server. Why should we call user@sip-proxy instead of using user@domain?

Remember that the later construction in addition to being user friendly, also adds redundancy, and with a proper DNS configuration also may add load balancing.


Created by oej, Last modification by oej on Tue 11 of Nov, 2003 [08:09 UTC]

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