Subject: [Sipping] SIP gateway between PSTN and IP network

From: Antonio Corghi <[email-masked]>
Date: Tue, 16 Nov 2004 12:18:04 +0100

Hello Everyone,

I'm trying to understand what should be the right call flow for a SIP
gateway between PSTN and IP networks. 
According to RFC3666, upon receiving the IAM message (let's suppose having
called party number +123456789), the SIP gateway should generate a SIP
Invite message whose "To" header field has the following syntax: 

sip:[email-masked];user=phone

where abc.com is known to be the domain of the outbound proxy used by the
SIP gateway to forward PSTN originated calls. 

But, if the gateway does know neither what's the domain of the outbound SIP
proxy (but only its IP address) nor what's the actual domain of the called
party, what's the right format of the To header field in the Invite message?
IHMO, two options are possible:

sip:+123456789@<IP_address_of_the SIP_outbound_proxy>;user=phone

or

tel:+123456789

Which one is better?

Thanks in advance
Bye,
	Antonio Corghi

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---

From: "Peterson, Jon" <[email-masked]>
Date: Tue, 16 Nov 2004 19:55:18 -0500

See RFC3824 (which explains how to use telephone numbers in SIP, and what a
SIP UA should do when confronted with a telephone number). The guidance
given in RFC3666 is good advice, but in cases where the advice cannot be
followed for various reasons, there are alternatives described in RFC3824.

In your case, I think it would be appropriate to create a tel URI and
forward the request to the known IP address of the outbound SIP proxy
server.

Jon Peterson
NeuStar, Inc.

> -----Original Message-----
> From: Antonio Corghi [mailto:[email-masked]]
> Sent: Tuesday, November 16, 2004 3:18 AM
> To: [email-masked]
> Subject: [Sipping] SIP gateway between PSTN and IP network
> 
> 
> Hello Everyone,
>