[cisco-voip] SIP as a gateway Protocol

Ryan Ratliff rratliff at cisco.com
Wed Nov 4 11:31:07 EST 2009


You define the SIP gateway in CUCM as a SIP trunk.  That's what he is  
referring to.

-Ryan

On Nov 4, 2009, at 11:26 AM, Voice Noob wrote:

Thanks for your response. You mentioned SIP trunks in your respoonse.  
Just to verify again I am NOT talking about a SIP trunk. I am talking  
at a SIP gateway.

On Wed, Nov 4, 2009 at 8:57 AM, Nick Matthews <matthnick at gmail.com>  
wrote:
I think there are a few different factors - but it's the protocol I
would use if I was administering my network.

We see a lot of SIP gateways, and it's definitely being deployed.

Some of the advantages:
-Easy to troubleshoot.  You can read up on SIP and learn the basics
2-3x faster than other protocols.  It's clear and concise for the most
part.
-Interop.  Most of the new devices coming out are all running SIP.
You'll have less pain with SIP-SIP than SIP-H323 or SIP-MGCP, plus
experience with it already.
-Easier transition to SIP as your PSTN connection (last post) if/when
you decide to make that jump.
-If you're already running H323, switching over is pretty easy.

Other considerations:
-H323 is still the best at video, and for a while, there doesn't
appear to be any real alternatives.
-MGCP is still the only 'centralized dial plan' protocol where you
don't have to do anything on your gateways at all.  If you're not good
with IOS and just 'want it to work', this is still the protocol to
look at.  It comes with it's own troubles, bugs, and instability
because of it.
-Some older devices don't support SIP yet, and you may still be
running H323 in the network anyways.
-For more advanced call flows and designs, you may run into some
unsupported features.  (Like using ANN for ringback, I think that is
still H323 only).
-I would recommend CUCM 6.x+ and 12.4(20)T and later for SIP trunks.
If you have older platforms like the 3700, CMM, that won't run 20.T, I
would stick to your existing protocol.  Likewise for CUCM versions
prior to 6.x.  The SIP stacks in the versions prior just aren't as
stable or have as many features.


-nick


On Wed, Nov 4, 2009 at 8:37 AM, VoiceNoob <voicenoob at gmail.com> wrote:
 > 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.
 >
 >
 >
 > 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.
 >
 >
 >
 > From: cisco-voip-bounces at puck.nether.net
 > [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of Tim Smith
 > Sent: Tuesday, November 03, 2009 10:07 PM
 > To: Nick Matthews
 > Cc: CiscosupportUpuck
 > Subject: Re: [cisco-voip] SIP as a gateway Protocol
 >
 >
 >
 > 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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 >>> cisco-voip mailing list
 >>> cisco-voip at puck.nether.net
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 >>
 >>
 >>
 >> --
 >>
 >> Cheers,
 >>
 >> Tim
 >>
 >>
 >> Sent from Sydney, Nsw, Australia
 >> _______________________________________________
 >> cisco-voip mailing list
 >> cisco-voip at puck.nether.net
 >> https://puck.nether.net/mailman/listinfo/cisco-voip
 >>
 >>
 >
 >
 > --
 >
 > Cheers,
 >
 > Tim
 >
 >

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