Hi there,
I am not sure this is the most appropiate list for
this matter of mine, but I'll give it a try. Do let me
know if I'm mistaken.
I am staring a project in which I will implement an
IMS architecture, with UA, HSS, CSCF, AS...
I have been checking out SIP Developing Tools found in
I noticed the example of Call Pickup in the SIP service examples draft
shows an INVITE Replaces message as part of the message flow. Could an
UPDATE also be used here and if so what are the pros/cons of both methods.
Do they in effect do the same thing ?
Thanks
John
Hi Cavan,
The time between 183 Session Progress and 200OK may be or may not be
controllable. It depends upon the time when user has picked up the phone. If
183 session progress was carrying someother media ( playing
tones/announcement etc), then it is the user response which will result in
200OK
Thanks for the clarification!
- Original Message -
From: Shawn Lewis [EMAIL PROTECTED]
To: Pong Cavan [EMAIL PROTECTED]; Singh, Indresh
[EMAIL PROTECTED]; sip-implementors@cs.columbia.edu
Sent: Thursday, May 19, 2005 6:01 PM
Subject: RE: [Sip-implementors] 183 Session Progress with SDP
From: Wainwright, John
I noticed the example of Call Pickup in the SIP service
examples draft
shows an INVITE Replaces message as part of the message flow.
Could an
UPDATE also be used here and if so what are the pros/cons of
both methods.
Do they in effect do the same thing ?
Since an
Hi John
UPDATE cannot be used as it is used to update session parameters, not to
replace a uri.
Regards
Ranjit
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wainwright,
John
Sent: Friday, May 20, 2005 6:33 PM
To: sip-implementors@cs.columbia.edu
Thanks for the info Wayne. It seems I need to get more info out of the call
originator. Thanks again!
- Original Message -
From: [EMAIL PROTECTED]
To: Pong Cavan [EMAIL PROTECTED]
Cc: Singh, Indresh [EMAIL PROTECTED];
sip-implementors@cs.columbia.edu;
[EMAIL PROTECTED]
Sent:
From: Jaikumar
Does this Proxy server support Presence?
Any presence-related SIP requests are proxied as are any other SIP requests.
But as it is unlikely that any UAs will be registered with this proxy,
presence features will not be informative.
Dale
Does this mean the UPDATE can only be used to modify e.g the codec ? Can it
also be used to modify the 'c=' part of the SDP to replace a call not yet
in the Answer state ?
John
-Original Message-
From: Avasarala Ranjit-A20990 [mailto:[EMAIL PROTECTED]
Sent: Friday, May 20, 2005 10:10
Hi
UPDATE can be used ot modify m= i.e codec and a= . Not c=
Also UPDATE can be used only after signalling is establishe, not prior to
that.
Regards
Ranjit
-Original Message-
From: Wainwright, John [mailto:[EMAIL PROTECTED]
Sent: Friday, May 20, 2005 8:58 PM
To: Avasarala
UPDATE can be used to modify m= i.e codec and a= .
Not c=
I do not think that statement is true. UPDATE can basically be used to
update everything a re-INVITE can update. This includes SDP changes
which comply with RFC3264.
If you think otherwise, please direct me towards statements within
Brett Tate wrote:
UPDATE can be used to modify m= i.e codec and a= .
Not c=
I do not think that statement is true.
I agree with Brett
UPDATE can basically be used to
update everything a re-INVITE can update. This includes SDP changes
which comply with RFC3264.
Not everything. The UPDATE
I got this error messages below: -
May 21 01:13:59 ser /usr/sbin/ser[25609]: WARNING:
upstream bug - 0-terminated packet
May 21 01:13:59 ser /usr/sbin/ser[25609]: ERROR:
parse_msg: message=
May 21 01:13:59 ser /usr/sbin/ser[25609]: ERROR:
receive_msg: parse_msg failed
May 21 01:14:59 ser
Yeah, sommebody else caught me on this too. My memory is just slipping.
3311 is clear that update *is* a target refresh request.
Sorry,
Paul
Brett Tate wrote:
UPDATE can basically be used to
update everything a re-INVITE can update.
This includes SDP changes which comply
with
So can I use UPDATE to modify all the SDP dialog including the c= field on
a call that has not been ACK'ed.
In particular I am thinking about UA--softswitch--Media Server sequence
which plays an announcement/collect a new DN from the UA forwards this info
to the softswitch ( in an application
Wainwright, John wrote:
So can I use UPDATE to modify all the SDP dialog including the c= field on
a call that has not been ACK'ed.
In particular I am thinking about UA--softswitch--Media Server sequence
which plays an announcement/collect a new DN from the UA forwards this info
to the softswitch
Thanks. I just omitted the reliable responses / PRACKS from my description
and hoped for the response indicating the sequence in general was OK.
John
-Original Message-
From: Paul Kyzivat [mailto:[EMAIL PROTECTED]
Sent: Friday, May 20, 2005 1:52 PM
To: Wainwright, John
Cc: 'Brett
From: Wainwright, John
In particular I am thinking about UA--softswitch--Media Server sequence
which plays an announcement/collect a new DN from the UA forwards this
info
to the softswitch ( in an application specific way ) who then sends out an
INVITE to the collected DN and UPDATES the
I like the simplicity of it but I have a couple of questions/comments.
1. I didn't want to make an assumption on the interface between the
softswitch and the media server - it does not have to be SIP
2 At what point would the SDP/dialog data be updated in the initial UA ? In
your sequence would
19 matches
Mail list logo
