[VoiceOps] High Quality, Reliable Voice via the Internet / SIPNOC

[Dan York]

Mark,

On Mon, Jun 9, 2014 at 2:50 PM, Mark R Lindsey <lindsey at e-c-group.com>
wrote:
>
>
> 2. Increase the ptime from 20 ms to 30-40 ms to reduce packet-drop exposure
>

A good number of years ago (it shocks me to realize it was probably about
10!) I was a product manager for SIP products at one of the IP-PBX vendors.
 I thought that we ought to be able to do better than having a ptime of
only 20 ms and so I did some digging.  I was very surprised to see the
sheer number of places where there were assumptions made that the ptime
would always be 20 ms.  In software... in hardware... in applications...
 not just from the vendor I was with but in the products of other vendors
as well.  It seemed like most everywhere SIP was deployed there was an
assumption of a 20ms ptime - and in many cases no way to use any other
value.

Now, obviously a great amount can change in 10 years - or not. (This *is*
telecom we're talking about!)   My point is really that this one might be
extremely hard to change in a way that would be widely useful and
interoperable. I realize others have raised technical concerns... mine is
more on the deployment side.  While I think it is interesting to explore, I
think part of that exploration should include whether changing the ptime is
something that can actually be done on any kind of realistic timeframe.

Just my 2 cents,
Dan


-- 

Dan York
dyork at lodestar2.com  +1-802-735-1624   Skype:danyork
My writing -> http://www.danyork.me/
http://www.danyork.com/
http://twitter.com/danyork
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