Re: [SR-Users] checking if calls are still active

2010-11-08 Thread Henning Westerholt
On Monday 01 November 2010, Vic Jolin wrote:
 But I heard that SST is only useful if the caller and callee supports it.
 is that true?

Hi Vic,

i think it works as well if only one of the both parties supports it.

Cheers,

Henning

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


Re: [SR-Users] checking if calls are still active

2010-11-01 Thread Sergey Okhapkin
SST.

On Monday 01 November 2010, Vic Jolin wrote:
 Hi,
 
 Just want to ask what is the best way to check if calls are still active
 especially when we are not handling media?
 
 Thanks
 


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


Re: [SR-Users] checking if calls are still active

2010-11-01 Thread Vic Jolin
But I heard that SST is only useful if the caller and callee supports it. is
that true?

On Mon, Nov 1, 2010 at 11:20 PM, Sergey Okhapkin
s...@sokhapkin.dyndns.orgwrote:

 SST.

 On Monday 01 November 2010, Vic Jolin wrote:
  Hi,
 
  Just want to ask what is the best way to check if calls are still active
  especially when we are not handling media?
 
  Thanks
 


 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


Re: [SR-Users] checking if calls are still active

2010-11-01 Thread marius zbihlei

On 11/01/2010 05:13 PM, Vic Jolin wrote:

Hi,

Just want to ask what is the best way to check if calls are still 
active especially when we are not handling media?


Thanks

Hello

There is SST (Sip Session Timers RFC 4028), which is implemented in the 
SST module, but at least 1 UA must know the extension. Also I have been 
working on a IETF draft  to use end-to-end OPTIONS ping to determine the 
operational status of a UA.(it works well with B2BUA, relatively well 
with stateful proxies). If you are interested I can talk more about the 
draft and how I propose using OPTIONS requests for this.(no changes on 
UAs are necessary)


Marius.

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


Re: [SR-Users] checking if calls are still active

2010-11-01 Thread Vic Jolin
That's interesting, can you tell me more about that draft?

On Mon, Nov 1, 2010 at 11:18 PM, marius zbihlei marius.zbih...@1and1.rowrote:

 On 11/01/2010 05:13 PM, Vic Jolin wrote:

 Hi,

 Just want to ask what is the best way to check if calls are still active
 especially when we are not handling media?

 Thanks

 Hello

 There is SST (Sip Session Timers RFC 4028), which is implemented in the SST
 module, but at least 1 UA must know the extension. Also I have been working
 on a IETF draft  to use end-to-end OPTIONS ping to determine the operational
 status of a UA.(it works well with B2BUA, relatively well with stateful
 proxies). If you are interested I can talk more about the draft and how I
 propose using OPTIONS requests for this.(no changes on UAs are necessary)

 Marius.


 ___
 SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
 sr-users@lists.sip-router.org
 http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users


Re: [SR-Users] checking if calls are still active

2010-11-01 Thread marius zbihlei

On 11/01/2010 05:24 PM, Vic Jolin wrote:

That's interesting, can you tell me more about that draft?


Hello

I have attached the draft.
There are 2 setups for using end-to-end OPTIONS ping

1. In Dialog... if one of the endpoints is a B2BUA, it can send at 
specific interval in-dialog OPTIONS ping (OPTIONS request, increasing 
cseq) to the other endpoint. Most well-behaved UAS will respond with a 
481 if the dialog not longer exists as per RFC 3261 .


 The response to an OPTIONS is
   constructed using the standard rules for a SIP response as discussed
   in Section 8.2.6.  The response code chosen MUST be the same that
   would have been chosen had the request been an INVITE.

Other error conditions are on a  408 , 404 etc (as any in-dialog request 
is not guaranteed to succeed).


This is already used some production environments.

2. Out of dialog. A proxy keeps sending OPTIONS request (no dialog 
involved here) to one of the UA as long as one of these 2 conditions exist:

a. the dialog has not ended via a method specific mean (BYE for INVITE etc)
b. a 408 (or other error codes) was received when generating a BYE.

These where the big two points of the draft.

Marius
On Mon, Nov 1, 2010 at 11:18 PM, marius zbihlei 
marius.zbih...@1and1.ro mailto:marius.zbih...@1and1.ro wrote:


On 11/01/2010 05:13 PM, Vic Jolin wrote:

Hi,

Just want to ask what is the best way to check if calls are
still active especially when we are not handling media?

Thanks

Hello

There is SST (Sip Session Timers RFC 4028), which is implemented
in the SST module, but at least 1 UA must know the extension. Also
I have been working on a IETF draft  to use end-to-end OPTIONS
ping to determine the operational status of a UA.(it works well
with B2BUA, relatively well with stateful proxies). If you are
interested I can talk more about the draft and how I propose using
OPTIONS requests for this.(no changes on UAs are necessary)

Marius.


___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
list
sr-users@lists.sip-router.org mailto:sr-users@lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users







Network Working Group M. Zbihlei
Internet-Draft 1and1 Internet AG
Expires: February 2, 2011August 2010



   Draft-mzbihlei-end-to-end-OPTIONS-ping-01

Abstract

   For VoIP providers, a common problem is related with finding the
   state of a dialog in certain error conditions that are caused by
   network problems, User Agent(UA) crashes etc.  This document
   describes a procedure for using the Session Initiation Protocol (SIP)
   OPTIONS method in order to allow a SIP entity to discover the status
   of a UA and decide the state of the dialog.

Status of this Memo

   This Internet-Draft is submitted in full conformance with the
   provisions of BCP 78 and BCP 79.

   Internet-Drafts are working documents of the Internet Engineering
   Task Force (IETF).  Note that other groups may also distribute
   working documents as Internet-Drafts.  The list of current Internet-
   Drafts is at http://datatracker.ietf.org/drafts/current/.

   Internet-Drafts are draft documents valid for a maximum of six months
   and may be updated, replaced, or obsoleted by other documents at any
   time.  It is inappropriate to use Internet-Drafts as reference
   material or to cite them other than as work in progress.

   This Internet-Draft will expire on February 2, 2011.

Copyright Notice

   Copyright (c) 2010 IETF Trust and the persons identified as the
   document authors.  All rights reserved.

   This document is subject to BCP 78 and the IETF Trust's Legal
   Provisions Relating to IETF Documents
   (http://trustee.ietf.org/license-info) in effect on the date of
   publication of this document.  Please review these documents
   carefully, as they describe your rights and restrictions with respect
   to this document.  Code Components extracted from this document must
   include Simplified BSD License text as described in Section 4.e of
   the Trust Legal Provisions and are provided without warranty as
   described in the Simplified BSD License.



Zbihlei Expires February 2, 2011[Page 1]

Internet-Draft End-to-end SIP OPTIONS Ping   August 2010


1.  Introduction

   Given the multitude of parties involved in a SIP Call, a common
   problem for providers is determining the status of the caller/callee
   during a dialog.  There are methods for making SIP enabled networks
   robust and efficient by providing redundancy at hardware or service
   level, but this is not the case for most UAs.  Hardware problems,
   software crashes, power failures are factors in the way a UA behaves,
   affecting