[cisco-voip] SIP as a gateway Protocol

Tue Nov 3 23:07:05 EST 2009

Hi Nick,

What about using SIP just as protocol to replace H323 / MGCP between CCM and
your Voice Gateway?

Cheers,

Tim

On Wed, Nov 4, 2009 at 2:46 PM, Nick Matthews <matthnick at gmail.com> wrote:

> You can get an over-the-top SIP provider, but if you get voice quality
> problems you'll have some trouble getting your ISP and SIP provider to
> play nicely.  Once it leaves your gateway you can't prove who may be
> causing the problem if there is jitter or packet loss.  Your ISP
> probably won't have any idea how to deal with it, because for
> traditional data these types of packet problems do not have much
> consequence.
>
> If you're cool with that, there are hundreds of providers of varying
> quality.
>
> The suggestion is still to go with the data line from the SIP
> provider.  You may be able to save some money on equipment
> consolidation or pricing depending on your volume / area as well.
> It's not the best scenario for every case, but there are certainly
> cases where it makes since and these cases are growing.
>
>
> -nick
>
> On Tue, Nov 3, 2009 at 9:32 PM, Tim Smith <thsglobal at gmail.com> wrote:
> > We dont have too many SIP providers here in Oz at the moment anyway.
> > We were talking about just using SIP between CCM and the Gateway. Vs MGCP
> > and H323.
> >
> > Fax / modem could definitely be a good point though.
> >
> > Cheers,
> >
> > Tim.
> >
> > On Wed, Nov 4, 2009 at 1:18 PM, Lelio Fulgenzi <lelio at uoguelph.ca>
> wrote:
> >>
> >> 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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> >
> >
> >
> > --
> >
> > Cheers,
> >
> > Tim
> >
> >
> > Sent from Sydney, Nsw, Australia
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>

-- 

Cheers,

Tim
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