[asterisk-users] Difference between SCCP and Cisco Call Manager traffic?

[Steve Dickey]

Call setup/teardown is handled with the SIP protocol while the actual call
audio is handled with RTP I think.  Check the config of your NAT devices
relative to RTP.

scd

On 4/16/07, shawnl at up.net <shawnl at up.net> wrote:
>
> I'm wondering about the difference between Cisco Call Manager and
> SCCP(2) network traffic.  I'm working on getting a Cisco 7960 phone to
> speak through a NAT to an asterisk box, without having to do a bunch of
> port
> forwarding on the NAT device.
>
> Without the nat, everything works fine.
>
> If the phone is behind a cisco pix that is doing the natting, it works
> fine (fixup protocol).
>
> If the phone is behind a more generic nat device, such as a linux box
> running ipfilter.  Then it can dial out, but there is no audio.  The
> interesting part is that this same phone, behind the same NAT works just
> fine if it is talking to a Cisco Call Manager box instead of an
> asterisk server.  So, I'm wondering what the difference in the protocols
> is
> (I no longer have access to the call manager box, so I can't look @ the
> traffic).  In a perfect world, I'd like to have the phone pretty much just
> work wherever it's plugged in as long as it can see the asterisk server.
>
>
> Any ideas ?
>
>
> Thanks
>
>
> Shawn
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-- 
Steve Dickey
Who is John Galt?
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