Hadriel,
I did not intend to suggest to use the registration interval timer for
keep-alive, of course all the regular flow keep-alive mechanisms of
outbound could be used.
The benefit is that we would reuse the efforts put in outbound, instead
of inventing yet another, only marginally different, mechanism for each
specific case. From the sending element point of view, sending a
REGISTER isn't very different from sending an OPTIONS; subsequent
keep-alives can then become STUN or double CRLF, until the registration
needs to be refreshed.
Using REGISTER instead of OPTIONS also provides the means to use the
standard authentication mechanisms, besides reusing outbound (and
there's a few other reusable things, like Path headers, but let's keep
it simple for the moment)
Regards,
Jeroen
Hadriel Kaplan wrote:
Hey Jeroen,
I'm not sure I follow you. The SIP Forum SIP-Connect profile, and now TISPAN,
make use of a REGISTER for an IP-PBX to register a whole set of AoR's to be
routed to. For those devices that need that ability, it makes sense to use
outbound of course, and its inherent keepalive, regardless of being behind a
NAT or not. And I don't think Christer's suggesting otherwise.
But for IP-PBX's or proxies which do not themselves represent an AoR target of
requests, or do not represent one in the domain of the device they want to
perform keepalive with, why would we want to make them add such registration
logic? What would be the gain over just simply sending OPTIONS at that point?
The beauty of a STUN or double-CRLF keepalive is, in my mind: it's trivial to
construct, trivial to parse, very small and fixed size, can be separated or
handled at a lower layer, explicit in its use, and does not get stopped by a
SIP-layer overload control. In short: it's a transport connection-layer
keepalive, no more no less. And indicating it in the Via keeps it a hop-link
thing, backwards-compatible, little or no provisioning, and with no URI target
addressing issues.
-hadriel
-----Original Message-----
From: Jeroen van Bemmel [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 25, 2008 5:56 PM
To: Christer Holmberg
Cc: Hadriel Kaplan; [EMAIL PROTECTED]; [EMAIL PROTECTED];
[EMAIL PROTECTED]; [email protected]
Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use
case
No, not a B2BUA. Each element would have a virtual UA function "in
parallel" to its regular function (e.g. being a proxy). And indeed, one
way to implement this would be to keep either one (one way) or two
(opposite ways) flows active between them.
Like so:
|-----------| |-----------|
| Proxy | | Proxy |
|-----------| |-----------|
| UA | <---------->| UA |
|-----------| |-----------|
Each "UA" would implement both a simple registrar and a UAC performing
registration. It could also be setup asymmetrically, with the (smaller)
IP-PBX doing a single registration towards the (bigger) Service provider
network. Registration expiry would denote a loss of connectivity.
Regards,
Jeroen
Christer Holmberg wrote:
Hi,
So, you are proposing that each element should be a B2BUA, and both
elements then register towards each other and use Outbound???
Regards,
Christer
-----Original Message-----
From: Jeroen van Bemmel [mailto:[EMAIL PROTECTED]
Sent: 25. kesäkuuta 2008 22:57
To: Hadriel Kaplan
Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED]
lucent.com; [email protected]; Christer Holmberg
Subject: Re: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy use
case
Hadriel, Markus,
Instead of standardizing keep-alives between proxies, how about we
define a "virtual UA" on each element (similar to the one described in
RFC3261 section 16.7 point 6) to be used to provide this functionality?
(using existing outbound functionality, perhaps both ways)
Regards,
Jeroen
Hadriel Kaplan wrote:
Yes I am of that same opinion - that any real "IP-PBX" or whatever big
enough NOT to be doing Registration, and to instead do static
provisioning or DNS, would be given a static hole/DMZ address in their
firewall/NAT. But some of my customers have told me otherwise.
(interestingly mostly in APAC region) There's also some concern that while
a static entry is there for inbound TCP connections, the PBX creates
outbound ones to the service provider which are ephemeral port sources and
need to live for very long durations (though why they can't just do TCP
keepalive is beyond me, but I'm no expert).
But anyway, the big issue we've seen is that we need both the PBX and
the service provider box to detect failure before an active call/request
attempt is made; to trigger alternate route selection without waiting for
transport failure, and as a method to detect liveness again and revert.
Today that's almost exclusively done with Options requests as far as I've
seen, and lots of people don't seem to like that.
-hadriel
-----Original Message-----
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 25, 2008 3:11 PM
To: [EMAIL PROTECTED]; Hadriel Kaplan;
[EMAIL PROTECTED]; [email protected]
Cc: [EMAIL PROTECTED]
Subject: RE: [Sip] Progress draft-holmberg-sip-keep: proxy-to-proxy
use case
Hi,
I'm a bit sceptical about the need for keep-alives between proxies.
It is of course entirely possible that an enterprise PBX is connected
to (or peering with) a service provider proxy through a NAT and/or a
firewall. However, wouldn't such a NAT or firewall be under the
administration of either the enterprise itself or its ISP (who quite
often would be the SIP service provider), and the required port
forwardings or firewall rules could be set through administration.
This means that there would not be need for keepalive traffic to
implicitely keep the mapping/pinhole open.
Or are there really deployment cases where there are SIP PBXs behind
unadministrated NATs or firewalls?
Wouldn't we then need keepalives for SMTP as well, or how has the
e-mail infrastructure managed to solve this problem?
Markus
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