Voice over IP Cisco - h323 (shuffle/hairpin) guru question

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Author h323 (shuffle/hairpin) guru question
Nick Marus

2006-11-13, 8:09 am

This might be a tuff one to explain...

I have callmanager 4.x at my core with cisco ip phones at remote sites
across my wan. Also, at these remote sites i have a non-cisco ip phone
system that has an ip trunk to the callmanager.

Everything is working fine, other than when the remote ip phone calls the
phone system at the same site, the rtp goes twice across the wan, once to
the callmanager, and then back again to the same same site and the ip phone
pbx.

I have verified the media path on both the cisco IP phone webpage and on the
non-cisco pbx.

Is there any way to shuffle (non hairpin) a call so that it does not
traverse the WAN? I have even placed a cisco ip2ip gateway in the mix at the
remote site and built the ip trunk to it from the non-cisco pbx, but the
call rtp still goes across the wan.

Any idea?

--
Nick Marus
nmarus@gmail.com

Ryan Ratliff

2006-11-13, 8:09 am

RTP should be terminated at both devices. Do you have 'MTP Required'
set on the IP gateway config in CCMAdmin? This would cause the RTP
to be terminated at the CM MTP resource (software by default) for all
calls involving that gateway.

If you require the MTP but don't want it to flow across the WAN then
simply configure a software MTP on a router at the remote site and
register it with the CM. Then configure the MRGL/MRG for the gateway
such that it uses that remote site MTP.

-Ryan

On Nov 8, 2006, at 11:26 AM, Nick Marus wrote:

This might be a tuff one to explain...

I have callmanager 4.x at my core with cisco ip phones at remote
sites across my wan. Also, at these remote sites i have a non-cisco
ip phone system that has an ip trunk to the callmanager.

Everything is working fine, other than when the remote ip phone calls
the phone system at the same site, the rtp goes twice across the wan,
once to the callmanager, and then back again to the same same site
and the ip phone pbx.

I have verified the media path on both the cisco IP phone webpage and
on the non-cisco pbx.

Is there any way to shuffle (non hairpin) a call so that it does not
traverse the WAN? I have even placed a cisco ip2ip gateway in the mix
at the remote site and built the ip trunk to it from the non-cisco
pbx, but the call rtp still goes across the wan.

Any idea?

--
Nick Marus
nmarus@gmail.com
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Nick Marus

2006-11-13, 8:09 am

Thanks! That did the trick.

On 11/8/06, Ryan Ratliff <rratliff@cisco.com> wrote:
>
> RTP should be terminated at both devices. Do you have 'MTP Required'
> set on the IP gateway config in CCMAdmin? This would cause the RTP
> to be terminated at the CM MTP resource (software by default) for all
> calls involving that gateway.
>
> If you require the MTP but don't want it to flow across the WAN then
> simply configure a software MTP on a router at the remote site and
> register it with the CM. Then configure the MRGL/MRG for the gateway
> such that it uses that remote site MTP.
>
> -Ryan
>
> On Nov 8, 2006, at 11:26 AM, Nick Marus wrote:
>
> This might be a tuff one to explain...
>
> I have callmanager 4.x at my core with cisco ip phones at remote
> sites across my wan. Also, at these remote sites i have a non-cisco
> ip phone system that has an ip trunk to the callmanager.
>
> Everything is working fine, other than when the remote ip phone calls
> the phone system at the same site, the rtp goes twice across the wan,
> once to the callmanager, and then back again to the same same site
> and the ip phone pbx.
>
> I have verified the media path on both the cisco IP phone webpage and
> on the non-cisco pbx.
>
> Is there any way to shuffle (non hairpin) a call so that it does not
> traverse the WAN? I have even placed a cisco ip2ip gateway in the mix
> at the remote site and built the ip trunk to it from the non-cisco
> pbx, but the call rtp still goes across the wan.
>
> Any idea?
>
> --
> Nick Marus
> nmarus@gmail.com
> ________________________________________
_______
> cisco-voip mailing list
> cisco-voip@puck.nether.net
> https://puck.nether.net/mailman/listinfo/cisco-voip
>




--
Nick Marus
nmarus@gmail.com

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