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Ill side with Dan on this one. its absolutely interesting to do a
10,20,40ms packet test out over dirty internet with only a few
endpoints involved but then to make that work with the whole
ecosystem and out to PSTN means you need to either:<br>
<br>
1: perform transrating at the edge on all calls to turn your 40ms
access side stream into 20ms that all your app servers will digest
inside the network and the PSTN will accept. This will require DSP
or transcoding resources of some kind for every stream<br>
<br>
or <br>
<br>
2: You will need to make sure anything the endpoint might want to
have a conversation with will support this 40ms stream. Good luck
when so much of this gear seems to be "write once, run forever,
patch never"<br>
<br>
<br>
I think the real answer here is in encapsulation and media
redundancy.<br>
<br>
<br>
<div class="moz-cite-prefix">On 6/9/2014 7:00 PM, Dan York wrote:<br>
</div>
<blockquote
cite="mid:CAOy=LPm7s6c=Qt7nJhwj0DRBheOeH2HfoYjtAB+XpFmx15am9A@mail.gmail.com"
type="cite">
<div dir="ltr">Mark,<br>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Mon, Jun 9, 2014 at 2:50 PM, Mark
R Lindsey <span dir="ltr"><<a moz-do-not-send="true"
href="mailto:lindsey@e-c-group.com" target="_blank">lindsey@e-c-group.com</a>></span>
wrote:
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
2. Increase the ptime from 20 ms to 30-40 ms to reduce
packet-drop exposure<br>
</blockquote>
</div>
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div>Just my 2 cents,</div>
<div>Dan</div>
<div><br>
</div>
<div><br>
</div>
-- <br>
<div
style="font-size:13px;color:rgb(136,136,136);font-family:arial,sans-serif"><br>
</div>
<div
style="font-size:13px;color:rgb(136,136,136);font-family:arial,sans-serif">Dan
York</div>
<div
style="font-size:13px;color:rgb(136,136,136);font-family:arial,sans-serif"><a
moz-do-not-send="true" href="mailto:dyork@lodestar2.com"
style="color:rgb(0,0,204)" target="_blank">dyork@lodestar2.com</a> <a
moz-do-not-send="true" value="+18027351624"
style="color:rgb(0,0,204)">+1-802-735-1624</a>
Skype:danyork</div>
<div
style="font-size:13px;color:rgb(136,136,136);font-family:arial,sans-serif">My
writing -> <a moz-do-not-send="true"
href="http://www.danyork.me/" target="_blank">http://www.danyork.me/</a></div>
<div
style="font-size:13px;color:rgb(136,136,136);font-family:arial,sans-serif"><a
moz-do-not-send="true" href="http://www.danyork.com/"
style="color:rgb(0,0,204)" target="_blank">http://www.danyork.com/</a></div>
<div
style="font-size:13px;color:rgb(136,136,136);font-family:arial,sans-serif"><a class="moz-txt-link-freetext" href="http://">http://</a><a
moz-do-not-send="true" href="http://twitter.com/danyork"
style="color:rgb(0,0,204)" target="_blank">twitter.com/danyork</a></div>
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<br>
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