Subject: [SIP] Operlap dialing within a SIP network

From: Christer Holmberg <[email-masked]>
Date: Mon, 05 Feb 2001 13:46:49 +0200

Hi,

The discussion about overlap dialing has been pretty much about calls
from SIP to PSTN, or PSTN-SIP-PSTN.

Consider the following scenario.

PSTN PHONE             GW           SIP SERVER      SIP UAS
-----------------------------------------------------------

--- 1 --->    -- IAM -> | -- INVITE -->  |
                        | <- 18x ------  | 
--- 2 --->    -- SAM -> | -- INFO ---->  |
                        | <- 200 (INFO)  |
                        | <- 18x ------  |
--- 3 --->    -- SAM -> | -- INFO ---->  |
                        | <- 200 (INFO)  |   -- INVITE
-->                                            
                        | <- 200 (INVITE)|   <- 200 ------ 

In this case, the SIP server (note that I am not using the word 'proxy')
must collect digits, sent from the PSTN side using overlap dialing, to
be able to find the SIP UAS, and INFO is used to provide the digits
after the INVITE has been sent.

My question is, should a SIP server (proxy, if you like) be allowed to
behave like this, or MUST it send 486s, and receive new INVITEs, until
it is able to find the UAS?

Regards,

Christer Holmberg
Ericsson Finland

---

From: Christer Holmberg <[email-masked]>
Date: Mon, 05 Feb 2001 14:14:21 +0200

I would like to clarify the scenario a little.

The SIP server is NOT a normal proxy in the SIP network, via which the
INVITE is routed to the UAS (if it was, we would use 486 responses until
the proxy can route the call to the UAS, like Tom proposed earlier in
the Operlap Dialing thread).

Instead, the server could be some kind of automatic phone switch, for
example connected to a VPN. The difference from a NAT would be that it
is connected directly to a number of UAS, so it needs the digits to be
able to find the correct UAS.

If the UAS would be, for example, a PSTN phone, the SIP server would be
a PSTN GW, so then it would be allowed, but what is it wants to use the
same logic to route a call to another network using SIP (or, maybe
another VoIP protocol).

Regards,

Christer Holmberg
Ericsson Finland

---

From: Christer Holmberg <[email-masked]>
Date: Tue, 06 Feb 2001 08:14:42 +0200

Hi Gonzalo,

I didn't say I am going to use INFO, but that is what has been proposed
earlier in this thread, so I assumed INFO would be used. The reason for
the discussion from the beginning was that the overlap signalling,
according to the SIP-ISUP mapping draft, may not work very well, and a
solution using 18x responses and INFO messages, instead of 484 and
re-INVITEs, was proposed.

Regards,

Christer Holmberg
Ericsson Finland

Gonzalo Camarillo wrote:
> 
> Hello Christer,
> 
> Why are you using INFO for sending extra digits?
> 
> Please, check the SIP-ISUP mapping draft, overlap signalling section.
> 
> Regards,
> 
> Gonzalo
> 
> Christer Holmberg wrote:
> >
> > Hi,
> >
> --
> Gonzalo Camarillo         Phone :  +358  9 299 33 71
> Oy L M Ericsson Ab        Mobile:  +358 40 702 35 35
> Telecom R&D               Fax   :  +358  9 299 30 52
> FIN-02420 Jorvas          Email :  [email-masked]
> Finland                   http://www.hut.fi/~gonzalo

---

From: Gonzalo Camarillo <[email-masked]>
Date: Tue, 06 Feb 2001 00:47:16 +0200

Hello Christer,

Regards,

Gonzalo

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