If you are not using any H323 trunks (like H323 trunk to ITSP or another
UCME) then you do not need the H323 Binding and Interface at all.

SCCP IP Phone are POTS Dial-peers, so when you make a call from SCCP IP
Phone to TDM PSTN, you can think of this call like a TDM to TDM call, where
you do not need any H323 or SIP interfaces.

IPPhone  <------sccp signalling ---------->  ip  x.x.x.x <------- analog
connection -------------> Telco
Where IP x.x.x.x is the IP address of telepony-service.

Almost the same goes the UCME SIP Phones, but instead of POTS dial-peers,
the SIP Phone are using VoIP Dial-peer. At this point, you have to consider
the SIP Bindings to an interface, where you will get the VoIP to TDM call.

IPPhone  <------SIP signalling ---------->  ip  x.x.x.x <------- analog
connection -------------> Telco
Where IP x.x.x.x is the IP address of voice register global

When using SCCP IP Phones, the RTP Stream is terminated at the
Telephony-Service interface, before it goes out to PSTN.
When using SIP IP Phones, the RTP Stream is terminated at the SIP interface
(Bind command), before it goes out to PSTN.






On Sat, Dec 31, 2011 at 6:05 PM, Ken Wyan <[email protected]> wrote:

> It is common to terminate PSTN line ( analog or ISDN) to the same CUCME
> router.
>
> I have seen many people saying it as an  h323 gateway  but we don't use
> any h323 commands in this router.
>
> I think following is the call flow. (signalling)
>
> IPPhone  <------sccp signalling ---------->  ip  x.x.x.x   <-------  h323
> signalling ------- >  ip y.y.y.y  <------- analog connection ------------->
> Telco
>
> ip x.x.x.x & ip y.y.y.y are defined as follows
>
> telephony-service
>  ip source-address x.x.x.x
>
> interface  a /a
>  ip address y.y.y.y  255.255.255.0
>  h323-gateway voip bind srcaddr y.y.y.y  ( normally this command is not
> used if CME is in the same router )
>  ip pim dense-mode  ( to send multicast moh to PSTN callers )
>
> Please confirm whether the above signalling path is correct or otherwise
> please correct me. (above is purely my assumption)
>
> Where does rtp stream terminate ( which ip )  before converting to TDM or
> analog ? Is it loopback interface ? Can we change it?
>
>
>
>
>
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