Is Genband's C20/CS2000 switches used as a gateway to PSTN and SIP network? or
is it used as a pure SIP network element (proxy, B2BUA, etc)?
If *only SIP is involved*, I can *not* figure out how the INFO method can be
used to convey overlapping digits. INFO is sent within an early dialog created
by the INVITE carrying the first dialed digits, if the INVITE is sent by the
caller's UAC, with who will it create the dialog? According to RFC3261, a
dialog is a relationship between UAs not between proxies or between a UA and a
proxy (right?).
Suppose the basic SIP trapezoid is used:
UAC->Proxy1->Proxy2->UAS
In a pure SIP network, if the caller UAC can create an early dialog with the
callee UAS, the digits in the Request-URI *is* complete! (subsequent digits
are not necessary because the UAS has been reached), so the UAC should *not*
create the dialog with the UAS, and then how should the dialog be created? and
between what elements?
thanks!
zhang
------------------ Original ------------------
From: "Joel Gerber"<[email protected]>;
Date: Tue, Jul 16, 2013 10:20 PM
To: "SIP Learner"<[email protected]>;
"sip-implementors"<[email protected]>;
Subject: RE: [Sip-implementors] Overlap signaling in a native SIP network
I know Genband's C20/CS2000 switches support (and default) to using the INFO
method for overlapping digits (this is what they call partial dialled digits
followed by subsequent digits).
Joel Gerber
Network Specialist
Network Operations
Eastlink
E: [email protected] T: 519.786.1241
-----Original Message-----
From: SIP Learner [mailto:[email protected]]
Sent: July-16-13 10:09 AM
To: Brett Tate; sip-implementors
Subject: Re: [Sip-implementors] Overlap signaling in a native SIP network
Thanks to all!
I found one internet draft that propose to use the INFO method to convey
subsequent dialed numbers:
http://tools.ietf.org/id/draft-zhang-sipping-overlap-01.txt
It claimed to resolve the issues related to the INVITE/484/ACK approach in
RFC3578, but this draft seems to be deceased only after one revision, don't
know what's wrong with it!
------------------ Original ------------------
From: "Brett Tate"<[email protected]>;
Date: Tue, Jul 16, 2013 07:56 PM
To: "SIP Learner"<[email protected]>;
"sip-implementors"<[email protected]>;
Subject: RE: [Sip-implementors] Overlap signaling in a native SIP network
> In my opinion, if only a SIP network is involved and no gateways are
> used, overlap signalling (e.g., the caller sends dialed digits to an
> outbound proxy in consecutive separate INVITEs for the outbound proxy
> to collect enough information and route the requests) is meaningless,
> because there are no physical connections to be established, am I
> right?
It isn't meaningless; it wastes network resources and the devices would need to
agree upon what should occur (i.e. how the digits are collected, et cetera).
Even though draft-ietf-bliss-shared-appearances provides a PUBLISH mechanism
for seizing an appearance, some vendors might also allow an INVITE/484/ACK
exchange to temporarily keep an appearance seized.
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