[asterisk-biz] Quote for feature: Check to see if a peer is up

[Alex Balashov]

That comes down to whether userspace, SIP stack-level OPTIONS pings are a "good estimate of RTT". :-)

--
This message was painstakingly thumbed out on my mobile, so apologies for brevity and errors.

Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Atlanta, GA 30030
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/, http://www.alexbalashov.com

On May 23, 2012, at 5:28 PM, "Olle E. Johansson" <oej at edvina.net> wrote:

> 
> 23 maj 2012 kl. 14:21 skrev Kevin P. Fleming:
> 
>> While this behavior is technically not RFC3261 compliant (and I've had discussions about it with at least one of the RFC's authors), it's quite useful in making decisions about whether a peer has become unavailable more quickly than would normally be possible. For a local peer that responds to OPTIONS requests in 100ms or less, if that peer stops responding, Asterisk will be able to make that determination in approximately 6 seconds, instead of the 32 seconds that would normally be required.
> 
> Form the RFC:
> 
> "T1 is an estimate of the round-trip time (RTT), and it defaults to 500 ms."
> 
> "The default value for T1 is 500 ms. T1 is an estimate of the RTT between the client and server transactions.
> Elements MAY (though it is NOT RECOMMENDED ) use smaller values of T1 within closed, private networks
> that do not permit general Internet connection. T1 MAY be chosen larger, and this is RECOMMENDED if it
> is known in advance (such as on high latency access links) that the RTT is larger. Whatever the value of T1,
> the exponential backoffs on retransmissions described in this section MUST be used."
> 
> I can't see how this is not RFC 3261 compliant. We have a good estimate of the RTT and use it.
> 
> /O
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