Thanks Nick, that is really great info!<br><br><div class="gmail_quote">On Thu, Nov 5, 2009 at 1:57 AM, 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;">
I think there are a few different factors - but it's the protocol I<br>
would use if I was administering my network.<br>
<br>
We see a lot of SIP gateways, and it's definitely being deployed.<br>
<br>
Some of the advantages:<br>
-Easy to troubleshoot. You can read up on SIP and learn the basics<br>
2-3x faster than other protocols. It's clear and concise for the most<br>
part.<br>
-Interop. Most of the new devices coming out are all running SIP.<br>
You'll have less pain with SIP-SIP than SIP-H323 or SIP-MGCP, plus<br>
experience with it already.<br>
-Easier transition to SIP as your PSTN connection (last post) if/when<br>
you decide to make that jump.<br>
-If you're already running H323, switching over is pretty easy.<br>
<br>
Other considerations:<br>
-H323 is still the best at video, and for a while, there doesn't<br>
appear to be any real alternatives.<br>
-MGCP is still the only 'centralized dial plan' protocol where you<br>
don't have to do anything on your gateways at all. If you're not good<br>
with IOS and just 'want it to work', this is still the protocol to<br>
look at. It comes with it's own troubles, bugs, and instability<br>
because of it.<br>
-Some older devices don't support SIP yet, and you may still be<br>
running H323 in the network anyways.<br>
-For more advanced call flows and designs, you may run into some<br>
unsupported features. (Like using ANN for ringback, I think that is<br>
still H323 only).<br>
-I would recommend CUCM 6.x+ and 12.4(20)T and later for SIP trunks.<br>
If you have older platforms like the 3700, CMM, that won't run 20.T, I<br>
would stick to your existing protocol. Likewise for CUCM versions<br>
prior to 6.x. The SIP stacks in the versions prior just aren't as<br>
stable or have as many features.<br>
<font color="#888888"><br>
<br>
-nick<br>
</font><div><div></div><div class="h5"><br>
<br>
On Wed, Nov 4, 2009 at 8:37 AM, VoiceNoob <<a href="mailto:voicenoob@gmail.com">voicenoob@gmail.com</a>> wrote:<br>
> Nick that is what I am asking. I in no way want to go with a SIP trunk to<br>
> the PSTN I just want to use SIP as my gateway protocol. So the Telco still<br>
> hands me a PRI / FXO lines and instead of using MGCP or H.323 I would use<br>
> SIP. As far as why drop H.323 I don’t have a reason to but when doing new<br>
> customer deployments I don’t want to put one thing in and then migrate to<br>
> something else two years down the road.<br>
><br>
><br>
><br>
> So I ask my question again has anyone used SIP as their GW protocol instead<br>
> of H.323? Any problems or things I should look for? Should I just not do it<br>
> yet.<br>
><br>
><br>
><br>
> From: <a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a><br>
> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net">cisco-voip-bounces@puck.nether.net</a>] On Behalf Of Tim Smith<br>
> Sent: Tuesday, November 03, 2009 10:07 PM<br>
> To: Nick Matthews<br>
> Cc: CiscosupportUpuck<br>
> Subject: Re: [cisco-voip] SIP as a gateway Protocol<br>
><br>
><br>
><br>
> Hi Nick,<br>
><br>
><br>
><br>
> What about using SIP just as protocol to replace H323 / MGCP between CCM and<br>
> your Voice Gateway?<br>
><br>
><br>
><br>
> Cheers,<br>
><br>
><br>
><br>
> Tim<br>
><br>
> On Wed, Nov 4, 2009 at 2:46 PM, Nick Matthews <<a href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>> wrote:<br>
><br>
> 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<br>
> 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>
><br>
><br>
> -nick<br>
><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.<br>
>>> We<br>
>>> had to buy a data package that was no more than 50% voice traffic. For<br>
>>> 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<br>
>>> 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<br>
>>> carrier,<br>
>>> it was a LOT of work. I understood it saved us quite a bit of money, so<br>
>>> 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<br>
>>> 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<br>
>>>> 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.<br>
>>>>> Any<br>
>>>>> experience with this? Can I get Calling Name and Number from the PSTN<br>
>>>>> 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>
>> Cheers,<br>
>><br>
>> Tim<br>
>><br>
>><br>
>> Sent from Sydney, Nsw, Australia<br>
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>><br>
><br>
><br>
> --<br>
><br>
> Cheers,<br>
><br>
> Tim<br>
><br>
><br>
</div></div></blockquote></div><br><br clear="all"><br>-- <br><br>Cheers,<br><br>Tim<br><br><br>Sent from Sydney, Nsw, Australia