[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. 
====================================================================== 
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 10:42 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 10:42  
---------------------------------------------------------------------- 
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               
====================================================================== 
05-27-08 10:42  file           Note Added: 0087356                          
======================================================================




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