Subject: RE: [SIP] Session-timer and REFER

From: "Venkatesh Venkataramanan" <[email-masked]>
Date: Wed, 21 Feb 2001 15:01:17 -0800

As far as I know you shd be able to modify session refreshes timeout
anytime, by sending a new INVITE with a higher/lower Session-Expires timeout
value.....

The second problem of the UA mistaking a refresh for a new call is not
possible, the To: tag in the INVITE would give away.....

-----Original Message-----
From: Christer Holmberg [mailto:[email-masked]]
Sent: Wednesday, February 21, 2001 2:30 PM
To: [email-masked]
Subject: [SIP] Session-timer and REFER

Hi,

More about the session-timer...

Assume user A establish a call with user B. A session-timer between A
and B is also establish.

Then, B wants to transfer A to user C, using REFER. A will then send
INVITE to C and establish a session. However, the session between A and
B is still open (even if there is no media) so the session-timer will
continue between them.

My first issue is that A (or B) may want to cancel the session-timer
during the session with C, and then maybe use it again if the media is
switched back to B at some stage. One solution would be for either A or
B
to increase the refresh timer during the A-to-C call, and then decrease
it
again if the A-to-B call continues, but is this possible?

The other issue is if B re-boots. Now, if A is sending the refreshes, B
will think it's a new call, so the phone will ring - AND, may start
sending media to B's RTP port specified in the SDP in the re-INVITE. So,
when A is talking to C, he may suddenly start hearing announcements,
ring tones, or even someone talking, from B. A's call control will not
know about this, since it will receive a 200 OK for the re-INVITE in
either case.

Comments?

Regards,

Christer Holmberg
Ericsson Finland

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---

From: Christer Holmberg <[email-masked]>
Date: Thu, 22 Feb 2001 01:11:23 +0200

Hi,

>possible, the To: tag in the INVITE would give away.....

Yes, but the idea of using re-INVITEs for the refreshes is because it is
supposed to provide
a keep-alive mechnism, to re-establish the call if a SIP node on the
path has rebooted
(or otherwise lost the call). So, even if there is a tag the UAS would
probably still
try to re-establish the call, because since it rebooted it doesn't
remember it had a
call which it transfered to another user.

Regards,

Christer Holmberg
Ericsson Finland