Subject: RE: [Sip] 3-way handshake bad!

From: "Pugaczewski, Jack" <[email-masked]>
Date: Mon, 17 Nov 2003 09:16:07 -0600

Question:

Why is a 3-way handshake bad ?  ISUP has used it successully for call
setup and call teardown:

Call Setup:    Setup -->; Call Proceeding <--; Connect <---
Call Teardown: Disconnect -->; Release <--; Release Complete -->

-----Original Message-----
From: [email-masked] [mailto:[email-masked]] On Behalf Of Paul
D.Smith
Sent: Monday, November 17, 2003 9:01 AM
To: [email-masked]'
Subject: [Sip] 3-way handshake bad!

All,

For context, see IETF 58 minutes.  There was a suggestion that the
non-INVITE avalanche problem be solved by using a 3-way handshake.  I
just want to flag that this idea can have unfortunate consequences
during congestion.

Firstly, consider a server under load.  The 3-way handshake requires the
following.

server processes request - #1
server sends response - #2
server processes request/handshake - #3.

By adding the 3rd step, we have _doubled_ the number of requests to be
handled by the server increasing the load and increasing the chances it
becomes congested.

As per my earlier e-mail, for TCP, the chances of the initial request
completing can be either 1.0 or 0.0 depending on the congestion.
Increasing the congestion at the server, increases the chance of
requests failing. For UDP, there is a 1/x (x >= 1) chance of the request
being received and processed by the server.  Similarly, there is a 1/x
chance of the 3rd request/handshake reaching the server.  This means the
chances of the 3-way handshake completing are 1/(x^2).  So adding the
third step greatly affects the chances of the 3-way handshake
completing.

Paul D.Smith
Network Protocols Group 
Data Connection Ltd (DCL)
Tel: +44 20 8366 1177  Email: [email-masked] 
Fax: +44 20 8363 1039  Web:   http://www.dataconnection.com

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_______________________________________________
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