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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: 0.41s
  • Memory usage: 2.17MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 2.68

SER module tm

TM module enables stateful processing of SIP transactions. The main use of stateful logic, which is costly
in terms of memory and CPU, is some services inherently need state. For example, transaction-based
accounting (module acc) needs to process transaction state as opposed to individual messages, and any
kinds of forking must be implemented statefuly. Other use of stateful processing is it trading CPU
caused by retransmission processing for memory. That makes however only sense if CPU consumption
per request is huge. For example, if you want to avoid costly DNS resolution for every retransmission
of a request to an unresolveable destination, use stateful mode. Then, only the initial message burdens
server by DNS queries, subsequent retranmissions will be dropped and will not result in more processes
blocked by DNS resolution. The price is more memory consumption and higher processing latency.

From user's perspective, there are two major functions :
   t_relay and  t_relay_to.

Both setup transaction state, absorb retransmissions from upstream, generate downstream retransmissions
and correlate replies to requests. t_relay forwards to current URI (be it original request's URI or a URI
changed by some of URI-modifying functions, such as sethost). t_relay_to forwards to a specific address.

In general, if TM is used, it copies clones of received SIP messages in shared memory. That costs the
memory and also CPU time (memcpys, lookups, shmem locks, etc.) Note that non-TM functions operate
over the received message in private memory, that means that any core operations will have no effect
on statefuly processed messages after creating the transactional state. For example, calling record_route
after t_relay is pretty useless, as the RR is added to privately held message whereas its TM clone is being forwarded.


Dependencies

TM depends on
  • No other SER module

Back to SIP Express Router
Created by oej, Last modification by utdrmac on Tue 14 of Jun, 2005 [19:56 UTC]

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