[asterisk-bugs] [Asterisk 0012725]: delayed RTP cause first sound to chop off
noreply at bugs.digium.com
noreply at bugs.digium.com
Tue May 27 10:42:53 CDT 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=12725
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Reported By: loloski
Assigned To:
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Project: Asterisk
Issue ID: 12725
Category: Core/RTP
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: SVN
SVN Branch (only for SVN checkouts, not tarball releases): 1.4
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
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Date Submitted: 05-26-2008 21:55 CDT
Last Modified: 05-27-2008 10:42 CDT
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Summary: delayed RTP cause first sound to chop off
Description:
snip dialplan
[from-sip]
exten => s,1,Answer
exten => s,n,AGI(agi.php)
exten => s,n,Hangup
snip agi.php
#!/usr/bin/php -q
<?php
ob_implicit_flush(false);
set_time_limit(30);
require 'class/phpagi.php';
error_reporting(E_ALL);
$agi = new AGI();
$agi->answer();
$pay_date=strtotime("2007-08-14"); /* Proof of concept code */
$agi->exec("SayUnixTime $pay_date||bdY");
$agi->hangup();
?>
When playing August 14, 2007, "august" has been chopped off and you will
hear only the audible "14" "2007" sound
a hack for this atm is to insert $agi->stream_file("silence/1"); before
calling
the SayUnixTime * apps
/* This is a workaround */
#!/usr/bin/php -q
<?php
ob_implicit_flush(false);
set_time_limit(30);
require 'class/phpagi.php';
error_reporting(E_ALL);
$agi = new AGI();
$agi->answer();
$pay_date=strtotime("2007-08-14"); /* Proof of concept code */
$agi->stream_file("silence/1");
$agi->exec("SayUnixTime $pay_date||bdY");
$agi->hangup();
?>
latency is not a problem at all since the latency between the nat UAC to
UAS is 40 ms with 0% packet loss codec involve gsm, uac is ekiga softphone
this is reproducible even without the agi script
I hope i made myself clear on this.
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file - 05-27-08 10:42
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It's not delayed, at least not from Asterisk. You can clearly see it
sending packets to your phone. What I'm thinking is happening is that until
you send a packet to Asterisk your router/NAT setup is discarding incoming
packets, causing audio to be lost.
Issue History
Date Modified Username Field Change
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05-27-08 10:42 file Note Added: 0087356
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