[cisco-voip] CUCM SIP To PSTN

[Nick Matthews]

Just a note on the CUCM prior to 8.6 MTP thing - this is pretty
commonly stated but isn't what I would recommend.

CUBE does delayed offer to early offer conversion (DO-EO) which will
convert delayed offer into early offer without needing an MTP.  That's
how people did it before CUCM 8.6 and while it's a nice feature, it
doesn't necessarily buy you much with SIP trunking since DO-EO
(early-offer forced) doesn't cost you anything or have any real
downsides.  If I was on 8.6 or later I would utilize that feature over
DO-EO conversion since it's cleaner, but not by much.

-nick

On Fri, Jun 15, 2012 at 11:20 AM,  <george.hendrix at l-3com.com> wrote:
> Currently running 8.5.  Getting ready to upgrade to 8.6.
>
> Bill Hendrix
>
>
> -----Original Message-----
> From: Heim, Dennis [mailto:Dennis.Heim at wwt.com]
> Sent: Friday, June 15, 2012 11:02 AM
> To: Hendrix, George (Bill) @ LSG - STRATIS; Nick Matthews; Adel Abushaev
> Cc: cisco-voip at puck.nether.net
> Subject: RE: [cisco-voip] CUCM SIP To PSTN
>
> So I would go SIP end to end. Plus using cube, keeps most of the complexities of the SIP configuration out of callmanager. I would go with a SIP gateway. What version of callmanager are you using? If you are prior to 8.6, you will need to allocate MTPs if your provides is doing early offer.
>
> Dennis Heim
> Sr. UC Engineer
> World Wide Technology
> Office: 314.212.1814
> Email: dennis.heim at wwt.com
> www.wwt.com
>
> -----Original Message-----
> From: cisco-voip-bounces at puck.nether.net [mailto:cisco-voip-bounces at puck.nether.net] On Behalf Of george.hendrix at l-3com.com
> Sent: Friday, June 15, 2012 9:46 AM
> To: Nick Matthews; Adel Abushaev
> Cc: cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] CUCM SIP To PSTN
>
>
>
> Basically, I currently have 2 gateways (2x 3925) and each have 4 PRIs (total of 8).  They are located on site with CUCM, so they are configured as MGCP gateways.  However, we are standing up a lot of other sites.  So I am thinking about switching these circuits to 2 10mb SIP circuits.  I would have increased call capacity and I could also work with the Telco to re-route calls destined to remote sites if their SIP circuit goes down to these circuits at the main office.  That's just something not offered with old PRI circuits, or at least no provider I've worked with has offered that with old PRI.  Then if I go with SIP, what's the best configuration in CUCM?  I haven't worked much with SIP circuits, but the ones I have worked with, the gateway was just an h.323 GW in CUCM.  Could it be added as a different type that would register with CUCM or is h.323 the best way to go?
>
> Appreciate any inputs...
>
> Regards,
>
> Bill Hendrix
>
>
> -----Original Message-----
> From: matthn at gmail.com [mailto:matthn at gmail.com] On Behalf Of Nick Matthews
> Sent: Thursday, June 14, 2012 10:17 PM
> To: Adel Abushaev
> Cc: Hendrix, George (Bill) @ LSG - STRATIS; cisco-voip at puck.nether.net
> Subject: Re: [cisco-voip] CUCM SIP To PSTN
>
> The scenario of:
> CUCM--SIP--GW--SIP--Provider
>
> The gateway magically turns into a CUBE, and there's a whole bunch of marketing and technical information on what happens when you do that.
>
> If you compare these two solutions:
> -CUCM direct SIP trunk to provider
> -Use MTP to keep media to a single IP
> -Or use NAT to keep internal addressing safe or -Use CUBE
>
> There's a whole bunch of scalability and troubleshooting problems that can arise from the first. Having a demarcation point at the GW (CUBE) is extremely helpful. As well, it prevents you from needing to NAT SIP which historically is a pretty terrible idea.  It also has some SIP security and flexibility options, and is a good centralization point for trunks.
>
> I haven't worked with anyone doing the direct trunk from CUCM to provider. Many providers are going to make their own rules which will include an SBC (industry term for CUBE).
>
> Short story - just use an SBC. There's about 3-4 compelling reasons.
>
> -nick
>
> On Thu, Jun 14, 2012 at 1:17 PM, Adel Abushaev <adel.abushaev at gmail.com> wrote:
>> You can set up a SIP trunk to your SP, assuming that you are a SIP
>> client of them.Otherwise, if you want to go over T1, then  you need to
>> terminate SIP on the GW to translate between SIP and ISDN PRI or
>> whatever other signalling you are using between you and telco.
>>
>> A.
>>
>> On Thu, Jun 14, 2012 at 5:44 AM,  <george.hendrix at l-3com.com> wrote:
>>> Hi everyone,
>>>
>>>
>>>
>>>   I know with h.323 gateways in CUCM, you have to configure dial
>>> peers on each gateway and for each CUCM.  Is it possible to create a
>>> SIP trunk directly from CUCM to the Telco?  Without having to do any
>>> dial peers on the gateway.  Or would it CUCM SIP <> SIP GW <> SIP
>>> PSTN, with SIP dial peers on the SIP GW for both CUCM and the PSTN?
>>>
>>>
>>>
>>> Thanks,
>>>
>>>
>>>
>>> Bill Hendrix
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> cisco-voip mailing list
>>> cisco-voip at puck.nether.net
>>> https://puck.nether.net/mailman/listinfo/cisco-voip
>>>
>>
>> _______________________________________________
>> cisco-voip mailing list
>> cisco-voip at puck.nether.net
>> https://puck.nether.net/mailman/listinfo/cisco-voip
>
> _______________________________________________
> cisco-voip mailing list
> cisco-voip at puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip