[asterisk-users] sip forking needed for ekiga 3.0

[SIP]

That strikes me as being careless and unreliable. Call me a purist, but 
I'm of the opinion that you should KNOW which interface to use based on 
which interface is registered and choose ONE interface based on the 
rules you've established 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? Or if you want to ensure 
a particular route based on why you created your multiple interfaces in 
the first place?

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."

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

N.


Brian J. Murrell wrote:
> On Thu, 2008-09-25 at 14:56 -0400, SIP wrote:
>   
>> Sending from multiple different points of origin doesn't make any sense
>> at all in either a logical or rational fashion. What's it supposed to
>> accomplish?
>>     
>
> It seems to be a "shot-gun" approach to making a SIP connection.  The
> assumption being I suppose that one or more of the IP aliases will fail
> for whatever reason (policy routing, filtering, etc.), so just try them
> all, and use the first one to make a completion and drop the others.
>
> b.
>
>   
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