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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=Section1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>SIP is like pre-draft 802.11n. Until SIP progresses back each vendor bucket of RFCs that they choose to support then it can be a standard based protocol. But for now there is Nortel SIP, Cisco SIP.<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'>I wish cisco would allow some of the POTS commands on the VOIP dial-peers such as forward digits etc. I know you can do it all with translation profiles/reg-ex but typing a command that resembles English would be appreciated.<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 style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:10.0pt;font-family:"Arial","sans-serif";color:#1F497D'>Dennis Heim<br>Network Voice Engineer<br>CDW Advanced Technology Services<br>11711 N. Meridian Street, Suite 225<br>Carmel, IN 46032<br><br>317.569.4255 Office<br></span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";color:black'>317.569.4201 Fax<br></span><span style='font-size:10.0pt;font-family:"Arial","sans-serif";color:#1F497D'>317.694.6070 Cell<o:p></o:p></span></p><p class=MsoNormal style='mso-margin-top-alt:auto;mso-margin-bottom-alt:auto'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><a href="mailto:dennis.heim@cdw.com" title="mailto:dennis.heim@berbee.com">dennis.heim@cdw.com</a><br><a href="http://www.berbee.com/" title="http://www.berbee.com/">www.berbee.com</a></span><span style='color:#1F497D'><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><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> Wednesday, November 04, 2009 5:08 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><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal style='margin-bottom:12.0pt'>Thanks Nick, that is really great info!<o:p></o:p></p><div><p class=MsoNormal>On Thu, Nov 5, 2009 at 1:57 AM, Nick Matthews <<a href="mailto:matthnick@gmail.com">matthnick@gmail.com</a>> wrote:<o:p></o:p></p><p class=MsoNormal>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><span style='color:#888888'><br><br>-nick</span><o:p></o:p></p><div><div><p class=MsoNormal><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>>>>>> ************************************<br>>>>>> This email may contain privileged and/or confidential information that<br>>>>>> is intended solely for the use of the addressee. If you are not the<br>>>>>> intended recipient or entity, you are strictly prohibited from<br>>>>>> disclosing,<br>>>>>> copying, distributing or using any of the information contained in the<br>>>>>> transmission. If you received this communication in error, please<br>>>>>> contact<br>>>>>> the sender immediately and destroy the material in its entirety,<br>>>>>> whether<br>>>>>> electronic or hard copy. This communication may contain nonpublic<br>>>>>> personal<br>>>>>> information about consumers subject to the restrictions of the<br>>>>>> Gramm-Leach-Bliley Act and the Sarbanes-Oxley Act. You may not<br>>>>>> directly or<br>>>>>> indirectly reuse or disclose such information for any purpose other<br>>>>>> than to<br>>>>>> provide the services for which you are receiving the information.<br>>>>>> There are risks associated with the use of electronic transmission.<br>>>>>> The<br>>>>>> sender of this information does not control the method of transmittal<br>>>>>> or<br>>>>>> service providers and assumes no duty or obligation for the security,<br>>>>>> receipt, or third party interception of this transmission.<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>>>>><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>>><br>><br>><br>> --<br>><br>> Cheers,<br>><br>> Tim<br>><br>><o:p></o:p></p></div></div></div><p class=MsoNormal><br><br clear=all><br>-- <br><br>Cheers,<br><br>Tim<br><br><br>Sent from Sydney, Nsw, Australia <o:p></o:p></p></div></body></html>