[cisco-voip] SIP as a gateway Protocol
Lelio Fulgenzi
lelio at uoguelph.ca
Tue Nov 3 21:18:22 EST 2009
>From our initial conversations with our PSTN providers, SIP was a few years away with feature parity with H323/MGCP/PRI trunks.
FAX support was definately out of the question, and there were crazy requirements about not being able to do voice only on the ethernet trunk. We had to buy a data package that was no more than 50% voice traffic. For us, we get our internet through our regional network at dirt cheap prices because we basically run a co-op. For others it might make sense to move to the same PSTN/SIP/Internet carrier, but for us it didn't. Even our backup internet link is cheaper than the PSTN provider could price I believe.
The other thing was route diversity and multiple demarcs. I think those were quite expensive where as now, we get it at no extra cost.
I've long been a proponent of if it ain't broke, don't fix it. Even when we went to tender and ended up switching our PRIs to another local carrier, it was a LOT of work. I understood it saved us quite a bit of money, so it was worth it in the end for a three year contract. That being said, don't expect that SIP will be cheaper than PRIs and/or without it's own problems.
Caveat Emptor as my friend Caesar said.
----- Original Message -----
From: Tim Smith
To: STEVEN CASPER
Cc: CiscosupportUpuck
Sent: Tuesday, November 03, 2009 8:46 PM
Subject: Re: [cisco-voip] SIP as a gateway Protocol
Also, SIP is slightly easier to troubleshoot than H323, much more so than MGCP. (And I also dont like MGCP anyway :)
Cheers,
Tim.
On Wed, Nov 4, 2009 at 12:45 PM, Tim Smith <thsglobal at gmail.com> wrote:
I like the idea.
More and more SIP trunks will be turning up. Why bother having to go from H323 to SIP. Simpler just to run SIP.
I also like SIP and how you can set it up to monitor the destination of your dial-peers. Shut them down if a CCM is down.
Cheers,
Tim
On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <SCASPER at mtb.com> wrote:
I assume you are talking traditional analog and digital PSTN gateways, why are you considering migrating to SIP to control these as opposed to H323? .
Steve
>>> Voice Noob <voicenoob at gmail.com> 11/3/2009 6:09 PM >>>
Has anyone started using SIP on the PSTN gateway? I want to use it instead of H.323 or MGCP and start migrating it to SIP on the gateway. Any experience with this? Can I get Calling Name and Number from the PSTN side?
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Cheers,
Tim
Sent from Sydney, Nsw, Australia
--
Cheers,
Tim
Sent from Sydney, Nsw, Australia
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