A to send 200
OK of Bye and ACK for 481 (Re-INV).
S to send 481.
See section 3.2.2 of RFC 5407.
Regards,
Dushyant P S Dhalia
Rancore Technologies,
Gurgaon, INDIA
Vishal Agrawal wrote:
Hi,
Assume that phone “A” issues a re-INVITE request to the SIP server “S” and
at the same time “S” iss
Hi Palie,
Thanks for the information.
BR,
Manoj
-Original Message-
From: Kamalakanta Palei (kpalei) [mailto:kpa...@cisco.com]
Sent: Thursday, December 10, 2009 8:06 AM
To: Manoj Priyankara [TG]
Cc: sip-implementors@lists.cs.columbia.edu; Rajani
Subject: RE: [Sip-implementors] difference
Hi Tomasz,
Greetings!
This must be a very useful post, but can not access this from the given
URL. If you got it with you, would you mind sharing it with the forum
members?
Thanks and regards,
Manoj
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-imple
Thanks Alok!
-Original Message-
From: Alok 2 Tiwari [mailto:alok2.tiw...@aricent.com]
Sent: Wednesday, December 09, 2009 11:46 PM
To: Manoj Priyankara [TG]; sip-implementors@lists.cs.columbia.edu
Subject: RE: [Sip-implementors] differences between Via, Record-Route
and Route headers
Hi M
"A" must honor BYE and send 200 OK.
Proxy "S" can send 481 or 403.
Regards,
Kamal
Cisco, Bangalore
India
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Vishal Agrawal
Sent: Thursday, December
Hi Manoj
Here I have put the example in detail how Record-Route, Route plays a
role in message routing and how it affects Req-URI.
Also a small description on Via.
Endpoint A sends a request say INVITE to proxy1, proxy1 forwards it to
proxy2 and proxy2 gives that INVITE to endpoint B.
Each proxy
IMO it would be inappropriate to send a 491 in this case, because it is
not glare, and it is perfectly ok to terminate the dialog. Given that it
has sent the BYE, I *think* it would be ok for S to send a 481 to the
reinvite. (But I haven't scrutinized the state machines regarding this.)
If not
Hi,
Assume that phone “A” issues a re-INVITE request to the SIP server “S” and
at the same time “S” issues a BYE request to “A”.
Is it valid for “A” to send a 491 response to this BYE request from “S”?
The SIP INVITE dialog is for a point to point media session.
Here is the quote from the Secti
El Miércoles, 9 de Diciembre de 2009, Dale Worley escribió:
> On Tue, 2009-12-08 at 00:56 +0100, Iñaki Baz Castillo wrote:
> > I just mean SIP/TEL URI's for two purposes:
> >
> > 1) For user/device SIP identity (I've never seen a SIP AoR containing
> > parameters).
> >
> > 2) The XUI field of XCAP
On Tue, 2009-12-08 at 23:42 +, Brez Borland wrote:
> Dale has a point. I was impressed watching talk dedicated to ipv6
> where major minds behind ipv6 development seemed to be straight
> ignorant, or least interested, in the notion of private network
> implementations such as NAT. their positio
On Tue, 2009-12-08 at 00:56 +0100, Iñaki Baz Castillo wrote:
> I just mean SIP/TEL URI's for two purposes:
>
> 1) For user/device SIP identity (I've never seen a SIP AoR containing
> parameters).
>
> 2) The XUI field of XCAP URI (XUI is the AoR of the user whose document we
> desire).
>
> So I
On Mon, 2009-12-07 at 17:30 -0500, pvall...@csc.com wrote:
> When will that happen to SBCs..:)
Actually, SIP works quite well with SBCs, as long as the SBC does not
attempt to restrict what features of SIP are used. Unfortunately, many
service providers configure their SBCs not only to perform s
Hi Manoj,
Some time ago I wrote a post that can give some info on this.
http://ictbackyard.com/archives/6
Kind regards,
- Tomasz Zieleniewski
--
ICT Backyard - http://ictbackyard.com
2009/12/9 Manoj Priyankara [TG]
>
> Dear All,
> Can anyone explain the differences
El Miércoles, 9 de Diciembre de 2009, Manoj Priyankara [TG] escribió:
> Dear All,
> Can anyone explain the differences between Via, Record-Route and Route
> headers? Further, are there any other routing related Headers associated
> with SIP?
RFC 3261. Just it.
--
Iñaki Baz Castillo
_
El Miércoles, 9 de Diciembre de 2009, mosbah.abdelkader escribió:
> Hello all,
>
> Has anyone tried to send or receive SIP and RTP over 80 port (HTTP port).
Both SIP and RTP over the same port?
This is not possible except if you use a VPN working at TCP level.
--
Iñaki Baz Castillo
__
Refer the below link for a detailed explanation of the header fields.
(Thanks to Jan)
http://www.openser.org/pipermail/users/2005-September/000839.html
Thanks & Regards,
Rajani
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists
Hi Manoj,
Via is used to route responses whereas Route and Record-Route headers are used
to route requests. User agents use Record-Route headers to build Route headers.
The other routing related headers are Path header (Refer RFC-3327) and
service-route header (Refer RFC-3608).
Thanks,
Alok Ti
Dear All,
Can anyone explain the differences between Via, Record-Route and Route headers?
Further, are there any other routing related Headers associated with SIP?
Thanks!
BR,
Manoj
___
Sip-implementors mailing list
Sip-implementors@lists.cs.columbia.ed
Hello all,
Has anyone tried to send or receive SIP and RTP over 80 port (HTTP port).
If yes, can you give a description of the isuues and problems discovered.
Thanks.
___
Sip-implementors mailing list
Sip-implementors@lists.cs.columbia.edu
https://list
Remember that there are two views of whether a dialog exists: one by the
UAC and one by the UAS. The goal is for them to agree, but there are
points in time when they don't agree.
So when the UAS sends a response < 300 with to-tag, *it* thinks there is
a dialog. This is true regardless of wheth
Hi Tom,
I might have exaggerated this. But yes NAT provides security by
explicitly isolating the network behind it. I guess the time when I
will be able to get, say a hundred IP addresses for personal use are
some good years away still. For SIP, I do believe that even today we
still have to accoun
Thanks a lot.
Regards,
Sunil
-Original Message-
From: Ritul Sonania [mailto:ri...@in.niksun.com]
Sent: Wednesday, December 09, 2009 2:35 PM
To: Sunil Bhagat (WT01 - Telecom Equipment)
Cc: sip-implementors@lists.cs.columbia.edu;
zhiqiang.z...@alcatel-sbell.com.cn
Subject: RE: [Sip-impleme
>
> Does this mean that there is no dialog?? Since 200 OK was lost in the
> network? Or would the UAS maintain dialog information based on 200 OK
> which it had sent... even though it was lost?
I suggest you read the RFC Section 13.3.1.4
The 2xx response is passed to the transport with an
Does this mean that there is no dialog?? Since 200 OK was lost in the
network? Or would the UAS maintain dialog information based on 200 OK
which it had sent... even though it was lost?
Regards,
Sunil
-Original Message-
From: Ritul Sonania [mailto:ri...@in.niksun.com]
Sent: Wednesday, De
Inline.
> -Original Message-
> From: sip-implementors-boun...@lists.cs.columbia.edu
> [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On
> Behalf Of Brez Borland
> Sent: woensdag 9 december 2009 0:43
> To: Dale Worley
> Cc: sip-implementors@lists.cs.columbia.edu
> Subject: Re: [S
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