[asterisk-bugs] [Asterisk 0017658]: CODEX SPEEX 16K

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=17658 
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Reported By:                celya
Assigned To:                
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Project:                    Asterisk
Issue ID:                   17658
Category:                   Codecs/codec_speex
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2010-07-16 03:03 CDT
Last Modified:              2010-07-16 11:45 CDT
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Summary:                    CODEX SPEEX 16K
Description: 
With the asterisk trunk, 16KHz speex codecs have a poor quality. It's
sounds like a frequency problem.
If you use SFLphone 0.9.8, when you establish a communication you have on
your screen "SPEEX/8000"


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 (0124617) celya (reporter) - 2010-07-16 11:45
 https://issues.asterisk.org/view.php?id=17658#c124617 
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My SIP account :
[34479]
type=friend
username=34479
insecure=very
qualify=no
nat=yes
host=dynamic
canreinvite=no
context=bbb-voip
disallow=all
allow=speex16
password=toto

In my context I juste use Playback(conf-placeintoconf)

The SDP is write :
sip show channel 209b23ea-0e8c-42fe-915b-2446aa2adb14 

  * SIP Call
  Curr. trans. direction:  Incoming
  Call-ID:                209b23ea-0e8c-42fe-915b-2446aa2adb14
  Owner channel ID:       SIP/34479-0000000a
  Our Codec Capability:   0x200000000 (speex16)
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   0x200000000 (speex16)
  Joint Codec Capability:   0x200000000 (speex16)
  Format:                 0x200000000 (speex16)
  T.38 support            No
  Video support           No
  MaxCallBR:              384 kbps
  Theoretical Address:    192.168.1.128:5060
  Received Address:       192.168.1.128:5060
  SIP Transfer mode:      open
  Force rport:            Yes
  Audio IP:               192.168.1.70 (local)
  Our Tag:                as42ab2675
  Their Tag:              e0ed6757-8519-46fa-8fb7-f6670cf7cc62
  SIP User agent:         
  Username:               34479
  Peername:               34479
  Original uri:           sip:34479 at 192.168.1.128:5060
  Caller-ID:              34479
  Need Destroy:           No
  Last Message:           Rx: ACK
  Promiscuous Redir:      No
  Route:                  sip:34479 at 192.168.1.128:5060
  DTMF Mode:              rfc2833
  SIP Options:            100rel replaces replace 
  Session-Timer:          Inactive

I try to change the SPEEX parameters in codecs.conf, but it's always the
same problem.

If I have two SIP account with SPEEX16 codecs, and juste a dial between
us, it the same problem, the sound is awful.

You can find my wireshark capture. 

Issue History 
Date Modified    Username       Field                    Change               
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2010-07-16 11:45 celya          Note Added: 0124617                          
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