Subject: [Sip] Tandem switch Call typing and routing

From: "Jenkal, Vijey" <[email-masked]>
Date: Wed, 13 Jul 2005 12:51:37 -0400

We are researching possible ways of performing PSTN call routing at a
Tandem switch that receives calls from an (or more) end-office switch
over a SIP interconnection. The tandem switch is responsible for
determining the type of the call and completing it to either the Local,
Local-toll or Long-distance PSTN trunk based on the call type. Based on
the current regulations, the call typing would have to be based on the
perspective of the originating switch rather than the tandem. Is there
any discussions in the SIP forum with regards to how SIP could be used
to convey the right information (either as call-type, rate-center
information or otherwise) to facilitate such call routing at the tandem?

The current alternatives that we are investigating are:

1. Find a standardized means of conveying the call-type information to
the tandem: The tandem switch would then use the combination of called
party number and call-type information to route the call to the
appropriate PSTN trunk. We are not, however, able to identify any
standard means of conveying this information.

2. Use a combination of calling and called party number information at
the tandem to perform the call typing: Our preliminary investigation
reveals that maintaining such a list of possible calling and called
party number information in the tandem switch's dial-plan would make the
dial-plan very complex and large. To this end, this alternative, while
still being looked into, does not look promising.

3. Use multiple SIP interconnections between the End-office and the
tandem switch: This architecture would emulate the current set up in the
PSTN network, where the end-office typically uses specific trunks to
forward call calls to the tandem base on the call type. The problem with
this approach, of course, is that in a market that has multiple
switches, there would be multiple SIP interconnections that would have
to be established and maintained, which we would like to avoid if
possible. The use of a single SIP interconnection between the switches
would make the implementation much more scalable and an improvement over
the current PSTN methodology.

We would be really interested in the forums feedback on the
alternatives. Are there other alternatives, from past discussions or
current drafts/RFCs that may be in progress, that we should look into as
well?

Thank you

Vijey Jenkal
VoIP Architecture Team P - 215-320-8259 E -
[email-masked]

_______________________________________________
Sip mailing list  https://www1.ietf.org/mailman/listinfo/sip
This list is for NEW development of the core SIP Protocol
Use [email-masked] for questions on current sip
Use [email-masked] for new developments on the application of sip

---

From: "Dale Worley" <[email-masked]>
Date: Wed, 13 Jul 2005 13:37:45 -0400

> From: Jenkal, Vijey
> Local-toll or Long-distance PSTN trunk based on the call type.

I'm assuming that what you want is for the end-office switch to classify the
call, but the tandem switch would be acting on that classification?

One method would be for the end-office switch, while it was working out how
to forward the call, to attach a URI parameter to the request-URI.  The
tandem would examine (and remove) the URI parameter to determine how to
route the call.

Dale

---

From: Dean Willis <[email-masked]>
Date: Thu, 14 Jul 2005 17:00:46 -0500

> information or otherwise) to facilitate such call routing at the 
RFC 3966 defines the tel: URL scheme, and it allows for a "context" 
parameter.

I suggest you have the edge switch set the "context" parameter to 
indicate its perspective, then have the tandem siwtch act on that 
context.

--
Dean