[asterisk-users] OT - Is sip.instance useful ?

[Olivier]

2008/10/2 SIP <sip at arcdiv.com>

> Olivier wrote:
> > Hi,
> >
> > I've seen some hardphones or Softswitchs now support this sip.instance
> > feature :
> >
> http://www.softarmor.com/wgdb/docs/draft-jennings-sipping-instance-id-01.txt
> >
> > I don't really see any convincing use of this draft but I would be
> > curious to share thoughts on it.
> >
> > Cheers
> >
> > ------------------------------------------------------------------------
> We see similar things a lot from X-Lite (although their implementation
> is somewhat different and has, in the past, been broken) -- using
> rinstance or some such. The IDEA is sound to a point. It's useful to be
> able to have an identifier with the UA to determine which UA is which
> for purposes of differentiation if people are logging in multiple times
> with the same username (something Asterisk doesn't allow).
>
> It's usually handled with simple registration parameters using IP/port
> combinations as differentiators, but if you're running a symmetric NAT,
> that may be misleading or even non-functional. The instance COULD act as
> an additional identifier to help clarify those situations on the SIP
> side (as opposed to just the RTP side).
>
> I do NOT, however, feel that such an identifier need last through
> reboots or even be semi-permanent in any way. It's logically nice to
> have a separation of instances of the same IP's clients (some of our
> users log in multiple times from multiple machines from the same IP) for
> programmatic purposes perhaps, but on a reboot, a new mapping should and
> would be sent via the REGISTER request, and so keeping this data across
> UA reboots seems.... unnecessary. And likely a security risk.
>
> Where are you seeing this crop up?


I've seen it in Thomson ST2030 hardphones naming compliance with Sylantro
software.
I read the mentioned draft RFC and wondered what it be useful for ?

If an hardphone has a SIP ID valued to its MAC address, it could remain
valid between reboots though I don't think the sip.instance's goal is just
to provide that ...
Maybe it's more relevant during other period of endpoint's liflecycle :
before a phone is configured with a stable ID ...



>
>
> N.
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081002/ceffb511/attachment.htm