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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Try SIP if at all possible. I have had
mixed results with IAX that SIP made go away. If you try SIP, you can at least
rule out IAX as the cause.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p><font size=2 color=navy face="Times New Roman"><span style='font-size:10.0pt;
color:navy'>Thanks,<br>
Steve Totaro<br>
<a href="http://www.asteriskhelpdesk.com">http://www.asteriskhelpdesk.com</a><br>
KB3OPB<br>
</span></font><font color=navy><span style='color:navy'> </span></font><o:p></o:p></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>Yossi Ben Hagai<br>
<b><span style='font-weight:bold'>Sent:</span></b> Saturday, April 28, 2007
4:41 AM<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [asterisk-users] Two
Connected Servers Sound Quailty</span></font><o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Hi Matt,<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>you didn't mention what type/bw of each site Internet connection, i
suggest that you try to split the scenario into smaller pieces:<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>- run long term pings between the server while you make a call and
check for packet loss.<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>- make internal calls between extensions on the same branch and verify
that both servers work okay (eliminating Internet connectivity)<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>- register a UA from one site to a server on the other site, make a
call and viceversa (eliminating a problem on one of the servers).<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>- check for speed/duplex setting on NIC and switch port.<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>- check if the sound quality issues are symmetric (does both sides
experience the sound cut or it only happens on a specific site).<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>- make sure you don't use G.711 as it consumes bw and from the codec
list you've mentioned has the lowest tolerance to packet loss.<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Since the problems are intermittent my bet is that someone in the
office is have the p2p client work overtime or sending lots emails with funny
attachments <br>
<br>
<o:p></o:p></span></font></p>
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<p class=MsoNormal><span class=gmailquote><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>On 4/28/07, <b><span style='font-weight:bold'>Matt
Gardner</span></b> <<a href="mailto:garddawg@gmail.com">garddawg@gmail.com</a>>
wrote:</span></font></span> <o:p></o:p></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Ok this is my first post and I will try to keep it short.<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>I have searched everywhere and haven't found an answer to my question<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>I have two Trixbox servers that are connected over the Internet via an
IAX2 connection. We are experiencing very poor sound quality.
I have tried many different codecs gsm, ilbc, g729, g711 and all seem
to have the same problem. (All though g729 seems to work the best but
still isn't reliable) The problems are intermittent sometimes the sound
will cut out for 3-4 seconds and other times the sound will just be loosing
every other word, and other times it sounds just fine. <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Also, we have been using Skype over this same Internet connection and
have very good sound quality with very few lost words.<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>So here are my questions.<o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>First, is it a correct assumption to say that because Skype works well
over this connection then I should be able to get asterisk to work over this
connect. I am hoping that Skype isn't "better" then asterisk in
this area. <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>If I should be able to get the same sound quality could you point me in
the right direction on how to achieve this. (I have tried messing with
the jitterbuffer but haven't been able to find very good docs on how to utilize
this functionality so about all I have done is set jitterbuffer=yes) <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'> <o:p></o:p></span></font></p>
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<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Thanks in advance.<o:p></o:p></span></font></p>
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<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'><br>
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