[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 07:31:23 CDT 2008
    
    
  
The following issue requires your FEEDBACK. 
====================================================================== 
http://bugs.digium.com/view.php?id=12725 
====================================================================== 
Reported By:                loloski
Assigned To:                
====================================================================== 
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:              
====================================================================== 
Date Submitted:             05-26-2008 21:55 CDT
Last Modified:              05-27-2008 07:31 CDT
====================================================================== 
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.
====================================================================== 
---------------------------------------------------------------------- 
 file - 05-27-08 07:31  
---------------------------------------------------------------------- 
Please provide an rtp debug and console output with this. Is the remote
side behind NAT? 
Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
05-27-08 07:31  file           Note Added: 0087328                          
05-27-08 07:31  file           Status                   new => feedback     
======================================================================
    
    
More information about the asterisk-bugs
mailing list