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<DIV><FONT size=2>ok thanks I understand now</FONT></DIV>
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style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><B>From:</B>
<A title=anthony.minessale@gmail.com
href="mailto:anthony.minessale@gmail.com">Anthony Minessale</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</A>
</DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Tuesday, March 23, 2010 1:52
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] rtp
timeout yet</DIV>
<DIV><BR></DIV>vad means voice activity detection meaning when you do not
speak it does not send rtp<BR>you can easily test that by running tshark on
your shell and see if the packets stop flowing when you don't
talk.<BR><BR>rtp-timeout-sec is how many sec to tolerate no rtp (it should be
a very high value because it's not madatory to send rtp and session timers is
a better way to make sure a call is still up)<BR><BR>rtp-hold-timeout-sec is
an alternate most likely even higher value to use while the call is in a held
state assuming you know there will be no audio during this time it's more
acceptable to have a higher number than the regular timeout.<BR><BR><BR><BR>
<DIV class=gmail_quote>On Tue, Mar 23, 2010 at 12:33 PM, Madovsky <SPAN
dir=ltr><<A
href="mailto:infos@madovsky.org">infos@madovsky.org</A>></SPAN> wrote:<BR>
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class=gmail_quote>
<DIV bgcolor="#ffffff">
<DIV><FONT size=2>No it doesnt' seem to use vad, unless you can call vad to
another standard name....</FONT></DIV>
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style="BORDER-LEFT: rgb(0,0,0) 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
<DIV class=im>
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV style="FONT: 10pt arial; BACKGROUND: rgb(228,228,228)"><B>From:</B>
<A title=anthony.minessale@gmail.com
href="mailto:anthony.minessale@gmail.com" target=_blank>Anthony
Minessale</A> </DIV></DIV>
<DIV class=im>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=freeswitch-users@lists.freeswitch.org
href="mailto:freeswitch-users@lists.freeswitch.org"
target=_blank>freeswitch-users@lists.freeswitch.org</A> </DIV></DIV>
<DIV>
<DIV></DIV>
<DIV class=h5>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Tuesday, March 23, 2010 1:09
PM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> Re: [Freeswitch-users] rtp
timeout yet</DIV>
<DIV><BR></DIV>your client probably has vad enabled which is a feature
that stops the rtp when you are not talking.<BR>one thing you can do is
disable or increase the duration of that rtp timeout on your sip profile
settings.<BR><BR><BR><BR><BR>
<DIV class=gmail_quote>On Tue, Mar 23, 2010 at 11:58 AM, Madovsky <SPAN
dir=ltr><<A href="mailto:infos@madovsky.org"
target=_blank>infos@madovsky.org</A>></SPAN> wrote:<BR>
<BLOCKQUOTE
style="BORDER-LEFT: rgb(204,204,204) 1px solid; MARGIN: 0pt 0pt 0pt 0.8ex; PADDING-LEFT: 1ex"
class=gmail_quote>Holly sh....! , I found the problem !<BR>I didn't know
that the fact I don't plug a mic on the legA and talk in it<BR>can
causes a rtp timeout !!!!<BR>so I have a question : what's happened if
in a conversation the legB talks<BR>talks and talks<BR>but legA
listens and don't say anything longer as the rtp timeout ? :D<BR>I tried
it, and indeed, there's an rtp timeout everytime I stop to talk.<BR>
<DIV><BR><BR><BR><BR>----- Original Message -----<BR>From: "Madovsky"
<<A href="mailto:infos@madovsky.org"
target=_blank>infos@madovsky.org</A>><BR></DIV>
<DIV>To: <<A href="mailto:freeswitch-users@lists.freeswitch.org"
target=_blank>freeswitch-users@lists.freeswitch.org</A>><BR></DIV>
<DIV>Sent: Tuesday, March 23, 2010 12:26 PM<BR>Subject: Re:
[Freeswitch-users] rtp timeout yet<BR><BR><BR></DIV>
<DIV>> thanks for your answer Jason.<BR>> yes the problem is yet
there. before<BR>> to submit a bug I want to be sure it's a mistake I
did.<BR>> so when I set inbound-bypass-media to true, the problem
disappear.<BR>> it's for sure a media problem.<BR>> A phone with a
public IP ( I tested on 6 PCs) registers well, but make a<BR>> rtp
timeout<BR>> A phone behind a firewall (tested on 6 PCs too)
registers well and doesn't<BR>> make rtp timeout.<BR>> I stopped
all IP rules like iptables etc... without success.<BR>> I'd like to
understand<BR>><BR>> Thanks<BR>><BR>><BR>><BR></DIV>
<DIV>> ----- Original Message -----<BR>> From: "Jason White"
<<A href="mailto:jason@jasonjgw.net"
target=_blank>jason@jasonjgw.net</A>><BR>> To: <<A
href="mailto:freeswitch-users@lists.freeswitch.org"
target=_blank>freeswitch-users@lists.freeswitch.org</A>><BR>>
Sent: Tuesday, March 23, 2010 4:32 AM<BR>> Subject: Re:
[Freeswitch-users] rtp timeout yet<BR>><BR>><BR>>> Madovsky
<<A href="mailto:infos@madovsky.org"
target=_blank>infos@madovsky.org</A>> wrote:<BR>>>> to be
sure I'm silly I downgraded from trunk 17001<BR>>>> and rtp
timeout disappeared.<BR>>><BR>>> Try the latest trunk. If
the problem still occurs, capture SIP traces and<BR>>> submit a
bug report, making sure the configurations are
identical.<BR>>><BR>>><BR>>>
_______________________________________________<BR></DIV>
<DIV>
<DIV></DIV>
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clear=all><BR>-- <BR>Anthony Minessale II<BR><BR>FreeSWITCH <A
href="http://www.freeswitch.org/"
target=_blank>http://www.freeswitch.org/</A><BR>ClueCon <A
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#freeswitch<BR><BR>FreeSWITCH Developer Conference<BR><A
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clear=all><BR>-- <BR>Anthony Minessale II<BR><BR>FreeSWITCH <A
href="http://www.freeswitch.org/">http://www.freeswitch.org/</A><BR>ClueCon <A
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