[asterisk-users] sip forking needed for ekiga 3.0

[Brian J. Murrell]

On Thu, 2008-09-25 at 15:31 -0400, SIP wrote:
> That strikes me as being careless and unreliable.

That's one argument.  I can also see the ekiga developers' argument
though and that's to strive for the most automatic functionality
possible.  The less things you have to ask users, the more likely you
are to "just work".

> Call me a purist, but 
> I'm of the opinion that you should KNOW which interface to use based on 
> which interface is registered

We are talking about IP aliases here, not real interfaces.

> and choose ONE interface based on the 
> rules you've established during registration.

What rules would you establish during registration?

> What happens if you want 
> to ensure that data goes across a VPN (in order to encrypt your VoIP 
> communications) instead of the public internet?

Presumably you have some [policy] routing that ensures that.  But on the
other hand, if you did have two addresses on an interface, one for VPN
and one for "everything else", unless you "shotgun" out you need to
either know which address to use or ask the user.  Either case may fail.

> That takes all the logic out of the equation and just says, "Here's a 
> bunch of packets. Figure out what to do with them. I'll be waiting for 
> your response."

I don't think it's quite that bad.  It's more like here's a bunch of
session requests, please complete them [you don't know it yet, but I'm
going to tear down all but the first one you complete].  But the glitch
is that even though I send you 3 of them, due to [policy] routing and
firewalling, you might only get one.

> There's a reason routing rules exist and mature services allow you to 
> control the interface from which it originates.

Really, I'm just the messenger here.  I doubt the ekiga team and the
asterisk team would be willing to sit down and discuss who is right
here, so I'm trying to be the conduit.

b.

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