[asterisk-bugs] [Asterisk 0010441]: RTCP Read Too Short generate strange DTMF tones on call.

noreply at bugs.digium.com noreply at bugs.digium.com
Tue Aug 28 10:35:58 CDT 2007


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=10441 
====================================================================== 
Reported By:                ygor
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   10441
Category:                   Core/RTP
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.10  
SVN Branch (only for SVN checkouts, not tarball releases):  1.4  
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             08-13-2007 13:24 CDT
Last Modified:              08-28-2007 10:35 CDT
====================================================================== 
Summary:                    RTCP Read Too Short generate strange DTMF tones on
call.
Description: 
Randomly Asterisk starts to issue messages like that:

[2007-08-13 15:33:54] WARNING[31849]: rtp.c:887 ast_rtcp_read: RTCP Read
too short

and the people hears some DTMF tones on the line.

We have detected that this occurs generally on Linksys PAP2-NA and Linksys
PAP2-T when using G729 codec.

Also I got this trouble on some IAX2 gsm trunks.

It appears to be randomly generated and sometimes it floods the CLI with
this message. ( generally when somebody inject DTMF tones on the call ).

This issue can be directly linked to the following Asterisk Forum post:
http://forums.digium.com/viewtopic.php?t=13114&highlight=&sid=949335ae5d5eddc10771bea0d7443599

Also from time to time, I get the "RTP Read Too Short" message too.

No jitterbuffers are used on my environment at the moment.
My asterisks are trunked on a voice-only network ( Motorola Canopy ).
There is absolutely no packet losses between the servers.
My environment is mixed, so I have SIP and IAX2 connections coming from
several places.
I generally use alaw, g729a and gsm codecs.
====================================================================== 

---------------------------------------------------------------------- 
 file - 08-28-07 10:35  
---------------------------------------------------------------------- 
What is the SIP device involved? 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
08-28-07 10:35  file           Note Added: 0069560                          
======================================================================




More information about the asterisk-bugs mailing list