SIP-H.323 Interworking Call Scenarios
vipin.palawat at wipro.com
Thu Mar 9 01:21:26 EST 2000
Thanks for your detailed comments.
We will come back to you with our opinions by today evening.
----- Original Message -----
From: Kundan Singh <kns10 at cs.columbia.edu>
To: Hemant Agrawal <hemantag at graffiti.net>
Cc: <schulzrinne at cs.columbia.edu>; <jdrosen at dynamicsoft.com>;
<korpim at sbs.de>; <karl.klaghofer at vs.siemens.de>; <rrroy at ATT.COM>;
<alan.johnston at wcom.com>; <steven.r.donovan at wcom.com>;
<Kevin.Summers at wcom.com>; <dean.willis at wcom.com>;
<henry.sinnreich at wcom.com>; VIPIN PALAWAT <vipin.palawat at wipro.com>;
<agboh at helios.iihe.ac.be>; <Robert.Sparks at wcom.com>;
<Chris.Cunningham at wcom.com>; <orit at radvision.com>;
<taylor at NORTELNETWORKS.COM>; <joon_maeng at vtel.com>;
<drwalker at ss8networks.com>; <paul.jones at TIES.ITU.INT>; <adilber at ATT.COM>;
<stephen.terrill at ericsson.com>
Sent: Wednesday, March 08, 2000 11:34 PM
Subject: Re: SIP-H.323 Interworking Call Scenarios
> Hi Hemant and Vipin,
> Its a very good collection of call flows' translation.
> Few comments:
> 1) Might want to include H.323---SIP---H.323 case where the
> call goes through two SGWs. Since, combining the
> two call flows (H.323-SIP, sec:1.1) and (SIP-H.323, sec:2.1)
> may not work. The SIP ACK in sec:1.1 has a different session
> description, which will trigger some "Mode Request"
> and/or "Open/Close LogicalChannel" procedures in
> 2) In sec:1.1,1.2,1.3; it might be better to send ACK
> immediately after receiving 200 OK, and send
> re-INVITE once H.245 procedures are complete.
> This way, we can avoid the retransmissions of 200 OK
> by SIP EP if H.245 procedures take more time to complete.
> Comments ??
> 3) In Fast Connect procedures (sec 3.1), I assume that
> if the normal H.245 procedure resumes after initial
> call setup (with faststart), and there is a change in
> H.323 side session description, then appropriate
> re-INVITE message is sent to SIP EP.
> 4) Sec:4 Call transfer may be renamed to (Blind) Call Tranfer.
> 5) Sec:4.1. RTP should be between SIP EP(B) and SIP EP(C)
> 6) Sec: 5.1, you may send "181 Call is being forwarded"
> response to SIP (A).
> Can SIP (A) reject the call forwarding here ???
> Who pays for the call if the forwarded call
> costs more (longdistance/international) ? Any
> pointer to this topic is appreciated.
> 7) The call forwarding may be initiated by the
> proxy (after receipt of 486 Busy Here) or by
> the SIP endsystem itself (by responding with 302 Moved Temporarily)
> if it is configured to do so. However, that will not
> affect the translation much.
> 8) Sec:5.3, typo; SIP User (A) should be (B) and (B) should be (C).
> Considering the wide variety of scenarios between
> H.323 and SIP, would it be reasonable to start with
> a simple call translation specification,
> provide a state machine/pseudo code for
> message handling by the SGW (signaling gateway)
> and then extend the state machine/pseudo code
> to handle all the non-trivial call scenarios.
> Once the basic translation specification is ready,
> profiles for extensions (e.g., blind transfer
> using GK, call forwarding using proxy,
> translating forking proxy behaviour to H.323, etc.)
> may be specified.
> Thus, the initial specification may cover following
> - Simple SIP--SGW--H.323 translation, with SGW
> independent of any GK or proxy.
> - Translating (blind) call transfer by endsystems.
> - Translating call forwarding by endsystems.
> - How to handle, SIP re-INVITEs (change
> in session description/or media transport address)
> and change in
> H.323 logical channel/mode request.
> - How to handle H.323 fast start.
> Later on other things may be added:
> - Call hold translation
> - Call transfer/forwarding by GK and/or proxy.
> - SIP--H.323--SIP and H.323---SIP---H.323 scenarios.
> - SGW coexiting with GK or proxy.
> Motivation for such a separation is to allow
> the implementors more freedom in deciding what
> configuration to choose, while still maintaining
> the basic translation framework.
> For example, call forwarding to voicemail may
> be handled by
> - the endsystem by sending 302 moved, after say 4 rings.
> - the proxy server on receipt of 486 busy here
> - the proxy server, forking one branch of the request to the voicemail
> system which accepts the call after, say, 15 seconds.
> It turns out that once we have the initial specification,
> (as mentioned in above list) the advanced scenarios
> are just a "special cases"/"combination of cases" of the
> initial scenarios.
> Kundan Singh http://www.cs.columbia.edu/~kns10
> On Wed, 8 Mar 2000, Hemant Agrawal wrote:
> > Hi All,
> > There was a continuous discussion on the different configuration for the
> > SIP - H.323 interworking.
> > In our opinion, it would be better if some basic features (like Call
> > Transfer, Call Hold and Call forwarding) is also added as part of the
> > discussion.
> > We are enclosing some call flow examples of SIP-H.323 interworking.
> > give your comments on these call flows. Our stress is on the call
> > and others are just informational. We have specified few issues also
> > to make these protocols interwork.
> > Kindly let us know if you agree to include the feature discussion also
> > part of the SIP - H.323 interworking. We are ready to put our efforts
> > the feature interworking.
> > Best Regards
> > Hemant Agarwal
> > Vipin Palawat
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