[asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?

[Olivier]

2008/1/9, Johansson Olle E <oej at edvina.net>:
>
>
> 9 jan 2008 kl. 02.48 skrev Raj Jain:
>
> > This issue of phone vendors not supporting OPTIONS according to RFC
> > 3261
> > often comes up on this list. Like Kevin Fleming said, an OPTIONS
> > request is
> > supposed to be responded in the same way as an INVITE. Almost all
> > SIP phone
> > vendors have construed OPTIONS as some kind of a keep-alive request,
> > which
> > is wrong.
> Which we do too, by the way. In worst case, maybe Asterisk has set
> this industry
> standard.
>
> OPTIONS is far to heavy in processing on the server side to be used
> for keep-alives. I'm  starting to see devices that use it for checking
> capabilities - the proper way. To do this properly, we will have to
> authenticate the OPTIONs request and match it with the proper peer/
> user to get the proper codec settings, ACLs and such.
>
> Since all versions of Asterisk use OPTIONs for NAT-keepalives, I'm a
> bit hesitant to fix this. It's a catch 22. I want to do it properly,
> but then the amount of processing for each OPTIONs request that we
> receive is going to be a bit too much. Maybe one could ask vendors to
> add a header to the  OPTIONs packet saying "this is just a keep-alive.
> Give me a 200 OK without any parsing and be happy, because I don't
> care about the reply."
>
> Linksys has a setting and use NOTIFY for Keep-alives, which also is a
> poor solution, but at least something we can just give an error
> response to without a lot of processing. There was a proposal for
> PING, but it never got anywhere.


Here  (http://www3.tools.ietf.org/html/draft-ietf-sip-outbound-11#page-11
§3.5.2) using STUN technique is recommended.
Do you foresee phone manufacturers to support this ?


/O
>
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