[Freeswitch-users] Conference and Speex Wideband audio quality problem

[Richard Alam]

Hi Everyone,

Hopefully, the answer to my problem isn't in the book as I've yet to
order it. :)

First off, I'm one ot the devs of BigBlueButton
(http://code.google.com/p/bigbluebutton/). We currently use Asterisk
for voice conference.

Leif Jackson made BBB work with FreeSWITCH. Now, from BBB, a user can
join a voice conference in FS. We have a java SIP client (based on
Red5Phone) that calls FS.

All is working well for Ulaw.

I am now trying to make Speex WB work since Flash Player (where our
client runs) supports Speex WB. Using Speex WB frees the java SIP
client from  transcoding between Nellymoser and Ulaw. The java SIP
client just takes the speex audio data from RTMP and sends it to FS
through RTP.

However, the sound quality is bad. It is choopy and robotic.

Let me describe what I have done and what I found from debugging so far.

I modified vars.xml to add the speex codec.
================
  <X-PRE-PROCESS cmd="set"
data="global_codec_prefs=speex at 16000h@20i,speex at 8000h@20i,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
  <X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=speex at 16000h@20i,PCMU,PCMA,GSM"/>


Setup the conference in dialplan/public.xml
==================
    <extension name="bbb_conferences">
      <condition field="destination_number" expression="^(7\d{4})$">
          <action application="answer"/>
          <action application="conference" data="$1 at wideband"/>
      </condition>
    </extension>

When I start up freeswitch, it looks like it's loading the speex codec
fine. Full console log is here http://pastebin.com/rnHfgQYj
===============
2010-07-30 16:04:10.620375 [NOTICE] switch_loadable_module.c:251
Adding Application 'AvoidingDeadlock'
2010-07-30 16:04:10.620658 [CRIT] switch_loadable_module.c:926 Error
Loading module /opt/freeswitch/mod/mod_g723_1.so
**/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file:
No such file or directory**
2010-07-30 16:04:10.620726 [CRIT] switch_loadable_module.c:926 Error
Loading module /opt/freeswitch/mod/mod_g729.so
**/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No
such file or directory**
2010-07-30 16:04:10.621189 [CRIT] switch_loadable_module.c:926 Error
Loading module /opt/freeswitch/mod/mod_amr.so
**/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No
such file or directory**
2010-07-30 16:04:10.621547 [CONSOLE] switch_loadable_module.c:944
Successfully Loaded [mod_ilbc]
2010-07-30 16:04:10.621618 [NOTICE] switch_loadable_module.c:185
Adding Codec 'iLBC' (iLBC) 8000hz 30ms
2010-07-30 16:04:10.621647 [NOTICE] switch_loadable_module.c:185
Adding Codec 'iLBC' (iLBC) 8000hz 20ms
2010-07-30 16:04:10.622043 [CONSOLE] switch_loadable_module.c:944
Successfully Loaded [mod_speex]
2010-07-30 16:04:10.622108 [NOTICE] switch_loadable_module.c:185
Adding Codec 'SPEEX' (Speex) 32000hz 20ms
2010-07-30 16:04:10.622135 [NOTICE] switch_loadable_module.c:185
Adding Codec 'SPEEX' (Speex) 16000hz 20ms
2010-07-30 16:04:10.622160 [NOTICE] switch_loadable_module.c:185
Adding Codec 'SPEEX' (Speex) 8000hz 20ms
2010-07-30 16:04:10.622513 [CONSOLE] switch_loadable_module.c:944
Successfully Loaded [mod_h26x]

When I call into the conference, it looks like mod_conference does not
use speex codec but L16 at 16000hz.
======
2010-07-30 16:05:56.874138 [DEBUG] sofia.c:4317 Channel
sofia/external/75115 at 192.168.0.166 entering state [completed][200]
2010-07-30 16:05:56.874138 [DEBUG] switch_core_session.c:647 Send
signal sofia/external/75115 at 192.168.0.166 [BREAK]
2010-07-30 16:05:56.874138 [DEBUG] switch_channel.c:2494
(sofia/external/75115 at 192.168.0.166) Callstate Change RINGING ->
ACTIVE
2010-07-30 16:05:56.874138 [NOTICE] mod_dptools.c:746 Channel
[sofia/external/75115 at 192.168.0.166] has been answered
EXECUTE sofia/external/75115 at 192.168.0.166 conference(75115 at wideband)
2010-07-30 16:05:56.875481 [DEBUG] mod_conference.c:5237 Raw Codec
Activation Success L16 at 16000hz 1 channel 20ms
2010-07-30 16:05:56.875481 [DEBUG] mod_conference.c:5282 Raw Codec
Activation Success L16 at 16000hz 1 channel 20ms
2010-07-30 16:05:56.877940 [DEBUG] switch_core_codec.c:116
sofia/external/75115 at 192.168.0.166 Push codec L16:10
tport_wakeup_pri(0x95bfd98): events IN


Is this the correct behavior? I've tried using xlite to connect to the
conference, and the audio is good. Also, when connecting from our
client,
the audio file that gets played by FS to notify how many users are in
the conference is good. It's my voice that's bad.

I tried to use the echo application instead of the conference. When I
call from the BBB client, the sound quality is good.

I appreciate any tips on how I can fix the problem or proceed to
figure out how to fix it.


Thank you very much.

Richard

-- 
---
BigBlueButton
http://www.bigbluebutton.org
http://code.google.com/p/bigbluebutton