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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
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TRIP

From the RFC 3219 :



This document presents the Telephony Routing over IP (TRIP). TRIP is a policy driven inter-administrative domain protocol for advertising the reachability of telephony destinations between location servers, and for advertising attributes of the routes to those destinations.
 
TRIP's operation is independent of any signaling protocol, hence TRIP can serve as the telephony routing protocol for any signaling protocol.

The Border Gateway Protocol (BGP-4) is used to distribute routing information between administrative domains. TRIP is used to distribute telephony routing information between telephony administrative domains. The similarity between the two protocols is obvious, and hence TRIP is modeled after BGP-4.

There is an open-source TRIP protocol stack as part of the Vovida VoIP suite of tools but it has not seen any work since 1.0.0 which was released in 2001. It currently does not compile on latest Linux 9.0 systems. It can be found on http://www.vovida.org/protocols/downloads/trip/

Both Acme Packet and Jasomi have TRIP-capable session controllers, though at this time not enough data is known about them to comment thoroughly. Additionally, these are hardware-based systems which have significant costs associated with their implementation.





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Created by oej, Last modification by paulej on Tue 19 of Oct, 2004 [07:59 UTC]

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