[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 10:33:17 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 10:33 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.
====================================================================== 

---------------------------------------------------------------------- 
 ygor - 08-28-07 10:33  
---------------------------------------------------------------------- 
Is this info useful ?!? (Notice the RTCP Read Too Short on the end!):

    -- Call accepted by 10.0.0.2 (format alaw)
    -- Format for call is alaw
    -- IAX2/GW1A-10 is proceeding passing it to IAX2/16102140-9
* Sent RTCP SR to 200.152.176.73:1232
  Our SSRC: 1234259135
  Sent(NTP): 1188316105.1284825088
  Sent(RTP): 240000
  Sent packets: 1500
  Sent octets: 30000
  Report block:
  Fraction lost: 0
  Cumulative loss: 0
  IA jitter: 0.0017
  Their last SR: 2136254054
  DLSR: 4.8270 (sec)

master*CLI> 

Got RTCP from 200.152.176.73:1232
PT: 200(Sender Report)
Reception reports: 1
SSRC of sender: 4210997425
NTP timestamp: 2981721.1717198848
RTP timestamp: 927033888
SPC: 1506I>     SOC: 30110
  Fraction lost: 27
  Packets lost so far: 137
  Highest sequence number: 34730
  Sequence number cycles: 0
  Interarrival jitter: 0
  Last SR(our NTP): 0.0000000000
  DLSR: 0.0000 (sec)
    -- IAX2/GW1A-17 is making progress passing it to
SIP/16102136-b3d10458
    -- IAX2/GW1A-10 is ringing
master*CLI> iax2 show pe
peers  peer   
[2007-08-28 12:48:30] WARNING[8255]: rtp.c:887 ast_rtcp_read: RTCP Read
too short
    -- Hungup 'IAX2/GW1A-17' 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
08-28-07 10:33  ygor           Note Added: 0069559                          
======================================================================




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