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<DIV><FONT face=Arial size=2>
<DIV><FONT face=Arial size=2>Hi Michel,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>how's beirut's weather with ya!</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>anyway, TTL stands for TIME To LIVE.</FONT></DIV>
<DIV><FONT face=Arial size=2>it's encapsulated on layer three of the OSI layer
to each packet going out that specific interface.</FONT></DIV>
<DIV><FONT face=Arial size=2>by default routers has a 16 TTL that means each
time the designated packet reaches a router (gets decapsulated) it gets a
-1...</FONT></DIV>
<DIV><FONT face=Arial size=2>this helps in preventing loops which would
eventually lead to congestion.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>now latency wise, for VOIP to operate correctly it
needs a latency of under 200 ms. (I currently have a microwave link , and
unfortunately im not getting that a latency less than 280 to my SIP
provider)</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>if your asterisk server is hosted online, you could
simply traceroute it and check the highest latency, point. and depending on
where that bottle neck would be, youll troubleshoot from there..</FONT></DIV>
<DIV><FONT face=Arial size=2>mine were on my ISP's international link, after
having a meeting with my account manager, I got my link routed through a
different international path which drastically decreased my
latency.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>now on a different approach, you absolutly have to
talk to your ISP/network administrator to provide you QOS for that specific IP
whether it's public or private.</FONT></DIV>
<DIV><FONT face=Arial size=2>depending on your network's traffic QOS would
surely help with no doubt.. this would decrease latency as well </FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>hope I've shed some light about this, if not well
the more knowledge the betteR</FONT></DIV>
<DIV> </DIV>
<DIV>best,</DIV>
<DIV>Roland</DIV></FONT></DIV>
<DIV style="FONT: 10pt Tahoma">
<DIV><BR></DIV>
<DIV style="BACKGROUND: #f5f5f5">
<DIV style="font-color: black"><B>From:</B> <A title=michofr@gmail.com
href="mailto:michofr@gmail.com">michel freiha</A> </DIV>
<DIV><B>Sent:</B> Monday, December 15, 2008 10:50 PM</DIV>
<DIV><B>To:</B> <A title=asterisk-users@lists.digium.com
href="mailto:asterisk-users@lists.digium.com">Asterisk Users Mailing List -
Non-Commercial Discussion</A> </DIV>
<DIV><B>Cc:</B> <A title=asterisk-users-bounces@lists.digium.com
href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</A>
</DIV>
<DIV><B>Subject:</B> Re: [asterisk-users] tcpdum</DIV></DIV></DIV>
<DIV><BR></DIV>
<DIV dir=ltr>Dear Sir,<BR><BR>There is no relation between TTL and the latency
on asterisk server?<BR><BR>Regards<BR><BR>
<DIV class=gmail_quote>On Mon, Dec 15, 2008 at 10:39 PM, Jeff LaCoursiere <SPAN
dir=ltr><<A href="mailto:jeff@jeff.net">jeff@jeff.net</A>></SPAN>
wrote:<BR>
<BLOCKQUOTE class=gmail_quote
style="PADDING-LEFT: 1ex; MARGIN: 0pt 0pt 0pt 0.8ex; BORDER-LEFT: rgb(204,204,204) 1px solid"><BR>TTL
is part of the UDP header (Time To Live). It isn't really about
the<BR>voice at all.<BR><BR>Length 345 is the number of bytes in the
packet.<BR><BR>j<BR><BR>On Mon, 15 Dec 2008, michel freiha wrote:<BR><BR>>
*Dear All,<BR>
<DIV>
<DIV></DIV>
<DIV class=Wj3C7c>> I run the below tcp dump on my asterisk
server<BR>><BR>> tcpdump -i eth0 -n -s0 -v udp port 5060<BR>><BR>>
I got the following result<BR>><BR>> 20:29:48.596867 IP (tos 0x10, ttl
64, id 0, offset 0, flags [DF], proto 17,<BR>> length: 373)
SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345<BR>><BR>> What
i need to know please what TTL means specifically and what is the best<BR>>
value og TTL and what is the lengh vale mean<BR>><BR></DIV></DIV>>
Regards*<BR>><BR><BR>_______________________________________________<BR>--
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