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.zbihlei@1and1.ro mailto:marius.zbihlei@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

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 sos@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