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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Nick that is what I am asking. I in no way want to go with a
SIP trunk to the PSTN I just want to use SIP as my gateway protocol. So the Telco
still hands me a PRI / FXO lines and instead of using MGCP or H.323 I would use
SIP. As far as why drop H.323 I don’t have a reason to but when doing new
customer deployments I don’t want to put one thing in and then migrate to
something else two years down the road. <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>So I ask my question again has anyone used SIP as their GW
protocol instead of H.323? Any problems or things I should look for? Should I
just not do it yet. <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>
cisco-voip-bounces@puck.nether.net [mailto:cisco-voip-bounces@puck.nether.net] <b>On
Behalf Of </b>Tim Smith<br>
<b>Sent:</b> Tuesday, November 03, 2009 10:07 PM<br>
<b>To:</b> Nick Matthews<br>
<b>Cc:</b> CiscosupportUpuck<br>
<b>Subject:</b> Re: [cisco-voip] SIP as a gateway Protocol<o:p></o:p></span></p>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal>Hi Nick,<o:p></o:p></p>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal>What about using SIP just as protocol to replace H323 / MGCP
between CCM and your Voice Gateway?<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal>Cheers,<o:p></o:p></p>
</div>
<div>
<p class=MsoNormal><o:p> </o:p></p>
</div>
<div>
<p class=MsoNormal style='margin-bottom:12.0pt'>Tim<o:p></o:p></p>
<div>
<p class=MsoNormal>On Wed, Nov 4, 2009 at 2:46 PM, Nick Matthews <<a
href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>> wrote:<o:p></o:p></p>
<p class=MsoNormal>You can get an over-the-top SIP provider, but if you get
voice quality<br>
problems you'll have some trouble getting your ISP and SIP provider to<br>
play nicely. Once it leaves your gateway you can't prove who may be<br>
causing the problem if there is jitter or packet loss. Your ISP<br>
probably won't have any idea how to deal with it, because for<br>
traditional data these types of packet problems do not have much<br>
consequence.<br>
<br>
If you're cool with that, there are hundreds of providers of varying quality.<br>
<br>
The suggestion is still to go with the data line from the SIP<br>
provider. You may be able to save some money on equipment<br>
consolidation or pricing depending on your volume / area as well.<br>
It's not the best scenario for every case, but there are certainly<br>
cases where it makes since and these cases are growing.<br>
<span style='color:#888888'><br>
<br>
-nick</span><o:p></o:p></p>
<div>
<div>
<p class=MsoNormal><br>
On Tue, Nov 3, 2009 at 9:32 PM, Tim Smith <<a
href="mailto:thsglobal@gmail.com">thsglobal@gmail.com</a>> wrote:<br>
> We dont have too many SIP providers here in Oz at the moment anyway.<br>
> We were talking about just using SIP between CCM and the Gateway. Vs MGCP<br>
> and H323.<br>
><br>
> Fax / modem could definitely be a good point though.<br>
><br>
> Cheers,<br>
><br>
> Tim.<br>
><br>
> On Wed, Nov 4, 2009 at 1:18 PM, Lelio Fulgenzi <<a
href="mailto:lelio@uoguelph.ca">lelio@uoguelph.ca</a>> wrote:<br>
>><br>
>> From our initial conversations with our PSTN providers, SIP was a few<br>
>> years away with feature parity with H323/MGCP/PRI trunks.<br>
>><br>
>> FAX support was definately out of the question, and there were crazy<br>
>> requirements about not being able to do voice only on the ethernet
trunk. We<br>
>> had to buy a data package that was no more than 50% voice traffic. For
us,<br>
>> we get our internet through our regional network at dirt cheap prices<br>
>> because we basically run a co-op. For others it might make sense to
move to<br>
>> the same PSTN/SIP/Internet carrier, but for us it didn't. Even our
backup<br>
>> internet link is cheaper than the PSTN provider could price I believe.<br>
>><br>
>> The other thing was route diversity and multiple demarcs. I think
those<br>
>> were quite expensive where as now, we get it at no extra cost.<br>
>><br>
>> I've long been a proponent of if it ain't broke, don't fix it. Even
when<br>
>> we went to tender and ended up switching our PRIs to another local
carrier,<br>
>> it was a LOT of work. I understood it saved us quite a bit of
money, so it<br>
>> was worth it in the end for a three year contract. That being said,
don't<br>
>> expect that SIP will be cheaper than PRIs and/or without it's own
problems.<br>
>><br>
>> Caveat Emptor as my friend Caesar said.<br>
>><br>
>><br>
>> ----- Original Message -----<br>
>> From: Tim Smith<br>
>> To: STEVEN CASPER<br>
>> Cc: CiscosupportUpuck<br>
>> Sent: Tuesday, November 03, 2009 8:46 PM<br>
>> Subject: Re: [cisco-voip] SIP as a gateway Protocol<br>
>> Also, SIP is slightly easier to troubleshoot than H323, much more so than<br>
>> MGCP. (And I also dont like MGCP anyway :)<br>
>><br>
>> Cheers,<br>
>><br>
>> Tim.<br>
>><br>
>> On Wed, Nov 4, 2009 at 12:45 PM, Tim Smith <<a
href="mailto:thsglobal@gmail.com">thsglobal@gmail.com</a>> wrote:<br>
>>><br>
>>> I like the idea.<br>
>>><br>
>>> More and more SIP trunks will be turning up. Why bother having to
go from<br>
>>> H323 to SIP. Simpler just to run SIP.<br>
>>><br>
>>> I also like SIP and how you can set it up to monitor the
destination of<br>
>>> your dial-peers. Shut them down if a CCM is down.<br>
>>><br>
>>> Cheers,<br>
>>><br>
>>> Tim<br>
>>><br>
>>> On Wed, Nov 4, 2009 at 12:25 PM, STEVEN CASPER <<a
href="mailto:SCASPER@mtb.com">SCASPER@mtb.com</a>> wrote:<br>
>>>><br>
>>>> I assume you are talking traditional analog and digital PSTN<br>
>>>> gateways, why are you considering migrating to SIP to
control these as<br>
>>>> opposed to H323? .<br>
>>>><br>
>>>> Steve<br>
>>>><br>
>>>> >>> Voice Noob <<a
href="mailto:voicenoob@gmail.com">voicenoob@gmail.com</a>> 11/3/2009 6:09 PM
>>><br>
>>>> Has anyone started using SIP on the PSTN gateway? I want to
use it<br>
>>>> instead of H.323 or MGCP and start migrating it to SIP on the
gateway. Any<br>
>>>> experience with this? Can I get Calling Name and Number from
the PSTN side?<br>
>>>><br>
>>>> ************************************<br>
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>>>> _______________________________________________<br>
>>>> cisco-voip mailing list<br>
>>>> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
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target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
>>>><br>
>>><br>
>>><br>
>>><br>
>>> --<br>
>>><br>
>>> Cheers,<br>
>>><br>
>>> Tim<br>
>>><br>
>>><br>
>>> Sent from Sydney, Nsw, Australia<br>
>><br>
>><br>
>> --<br>
>><br>
>> Cheers,<br>
>><br>
>> Tim<br>
>><br>
>><br>
>> Sent from Sydney, Nsw, Australia<br>
>><br>
>> ________________________________<br>
>><br>
>> _______________________________________________<br>
>> cisco-voip mailing list<br>
>> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
>> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip"
target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
><br>
><br>
><br>
> --<br>
><br>
> Cheers,<br>
><br>
> Tim<br>
><br>
><br>
> Sent from Sydney, Nsw, Australia<br>
> _______________________________________________<br>
> cisco-voip mailing list<br>
> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip"
target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a><br>
><br>
><o:p></o:p></p>
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<p class=MsoNormal style='margin-bottom:12.0pt'><br>
<br clear=all>
<br>
-- <br>
<br>
Cheers,<br>
<br>
Tim<br>
<br>
<o:p></o:p></p>
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