[cisco-voip] Gatewaying between SIP and H.323

[Steve Ames]

----- Original Message ----- 
> From: "Walenta, Phil" <philip.walenta at berbee.com>
>

> Cisco gateways actually can allow re-termination of H.323 calls to another
H.323 leg.  It's the IP Gateway to IP Gateway feature.

__Yah. Saw that on their site. Also saw reference to SIP support for
IP-IP Gateway in 2004 (and its 8 months through 2004 already). Is
this the only way to gateway between SIP and H.323? I'm bummed that
the obvious solution didn't just work... Thoughts? Will happily
take product recommendations as well. I'd really like to get this
going with something more scalable than HDV/PRI cards (not to
mention the voice quality is pretty ugly when you do that many
conversions (PSTN->SIP->PSTN->H323). Ick.

-Steve
______________________________

From: cisco-voip-bounces at puck.nether.net on behalf of Klaus Darilion
Sent: Wed 9/1/2004 5:36 PM
To: Steven E. Ames
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Gatewaying between SIP and H.323



ASFAIK this is not supported. The feaure is called "hairpin of media"
and means that the call leaves the GW on the "same side" it is received
eg: PSTN - PSTN or VoIP - VoIP.

hairpin of media is supported on the POTS side, but not on the VoIP
side. Therefore SIP-H323 is not possible.

Klaus

PS: This information is from the Cisco website - maybe its out-dated.

Steven E. Ames wrote:

> Hey. I'm hitting a bit of a snag. The solution should be fairly obvious
but I'm not seeing it. I'm running a little test using a Cisco 2651XM (IOS
12.2.13(T8)). Here's the scenario.
>
> A lot of SIP calls come into this box. All of the calls are prefixed 001
and are seen by the following incoming dial-peer:
>
> dial-peer voice 50 voip
>     application session
>     incoming called-number 001.T
>     session protocol sipv2
>     session target sip-server
>     fax rate disable
>     no vad
>
> So far so good. Normally I terminate these calls onto our TDM network via:
>
> dial-peer voice 5 pots
>     application session
>     destination-pattern 001.T
>     port 1/0:23
>
> And all is good with the world... now we have an H.323 gatekeeper and I
use this same Cisco box as a PSTN gateway for the H.323 phones. That also
works just fine. I have no problem at all calling from SIP<->TDM or
H.323<->TDM. However... what I'd like to do is have incoming SIP calls
delivered directly to H.323 phones (via gatekeeper and RAS).
>
> My first thought was fairly obvious:
>
> dial-peer voice 5 voip
>     application session
>     destination-pattern 001.T
>     session target ras
>
> But that fails quite miserably. The dial-peer is matched but that's the
end of it. Now I have two theories on why this fails:
>
> 1. I can't gateway SIP to H.323 with my hardware/IOS. This would be sad
but if someone can say its so and maybe point out options?
>
> 2. ras is attempting to match 001T instead of just T (e.g. 0012125551212
instead of 2125551212)? I'm under the, potentially mistaken, impression that
a destination-patter of 001.T will strip the 001. So maybe RAS just isn't
matching.
>
> Those are my two best guesses. I'm unclear on #2 because a statement such
as:
>
> num-exp 0012125551212 2125551212
>
> and changing destination-pattern to '2125551212' doesn't help at all.
>
> The closest I've come to getting this to work is routing the SIP call to
my TDM network and then bouncing it back to the Cisco and then to the H.323:
>
> SIP -> CISCO -> TDM -> CISCO -> H.323            works
> SIP -> CISCO -> H.323                                            doesn't
work
>
> Help?
>
> -Steve
>
>
>
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>
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