Hi Nick,<div><br></div><div>What about using SIP just as protocol to replace H323 / MGCP between CCM and your Voice Gateway?</div><div><br></div><div>Cheers,</div><div><br></div><div>Tim<br><br><div class="gmail_quote">On Wed, Nov 4, 2009 at 2:46 PM, Nick Matthews <span dir="ltr"><<a href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">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>
<font color="#888888"><br>
<br>
-nick<br>
</font><div><div></div><div class="h5"><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>
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>>><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>
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><br>
><br>
><br>
> --<br>
><br>
> Cheers,<br>
><br>
> Tim<br>
><br>
><br>
> Sent from Sydney, Nsw, Australia<br>
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><br>
</div></div></blockquote></div><br><br clear="all"><br>-- <br><br>Cheers,<br><br>Tim<br><br><br>
</div>