Does the latency remain more or less the same regardless of the
bandwidth load on the pipe?
If so, TOS bits (what you refer to as QoS) won't help you. You've
either got network issues (very likely if you have an intra-network ping
of 30 ms) or the outside host you're sending the traffic to
Stephen Reese wrote:
Does the latency remain more or less the same regardless of the
bandwidth load on the pipe?
If so, TOS bits (what you refer to as QoS) won't help you. You've
either got network issues (very likely if you have an intra-network ping
of 30 ms) or the outside host you're
On Oct 19, 2008, at 1:21 AM, Alex Balashov wrote:
Stephen Reese wrote:
Does the latency remain more or less the same regardless of the
bandwidth load on the pipe?
If so, TOS bits (what you refer to as QoS) won't help you. You've
either got network issues (very likely if you have an intra-
Alex is correct. Always check thereare no half-duplex links in your
path. If you have an older dsl/cable modem or router that only has a
10M ethernet, it is probably half. Also make certain there are no hubs
in the path. Keep in mind that colissions ar NORMAl for a hlaf duplex
connection. TCP
On Sun, Oct 19, 2008 at 12:31 AM, Stephen Reese [EMAIL PROTECTED] wrote:
My latency is kind of high and the voice delay is noticeable.
Then pretty much all you can do is lower the latency to lower the
voice delay, or use a connection to th e PSTN that has a marginally
lower delay if you have no
Does the latency remain more or less the same regardless of the
bandwidth load on the pipe?
If so, TOS bits (what you refer to as QoS) won't help you. You've
either got network issues (very likely if you have an intra-network ping
of 30 ms) or the outside host you're sending the traffic to is
Ping... =)
From: Luis Antonio Prata Barbosa [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: [asterisk-users] Latency, Jitter
On Fri, 2007-08-31 at 15:51 -0300, Luis Antonio Prata Barbosa wrote:
Does anybody know any software that give me Latencty, Jitter and Lost
packets to analyze my Call quality ???
The packet sniffer called Wireshark has a great RTP analysis tool that
will show you (and even graph!) the latency,
That would depend heavaly on your netowrk. Would your Swtiches (not
routers as TMDoE is layer 2) I pulled up an old posting from Mark on
TDMoE.
http://www.marko.net/asterisk/archives/0301/0566.html
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
On Fri, Oct 07, 2005 at 02:02:06PM +1000, Rod Bacon wrote:
Upon closer inspection, I don't think my system ever tries to establish a
zaptel native bridge. Is there somewhere where this function is
enabled/disabled?
Yeah, if you have echocancelwhenbridged or any other options that would
make
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Rod Bacon
Sent: Thursday, October 06, 2005 8:36 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Latency on bridged PRI calls
Nobody has been able to answer this. Not even Digium at
Upon closer inspection, I don't think my system ever tries to establish a zaptel
native bridge. Is there somewhere where this function is enabled/disabled?
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria,
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