[teknoids] Outcome: Classcaster teleconference webcast

Patricia Baia PBaia at mail.als.edu
Wed Feb 1 14:29:42 EST 2006


Great work Elmer!!!
 
I do have to say, currently this audio clip seems to be my favorite :) 
LOL... What is this conversation?   -Patty  
http://classcaster_news.classcaster.org/blog/resource/podcast/download/WS_10009.mp3
 
 
Patricia L. Baia  M.S.
Instructional Technologist & Ph.D Student
Albany Law School
518.445.3301
pbaia at mail.als.edu

>>> emasters at cali.org 2/1/2006 2:18:30 PM >>>

Well, it worked:)  Rosa Peters, Michael Samson, and John Mayer joined
me 
via conference call for a lively discussion that ended up lasting
nearly 
an hour.  The call was streamed live over the Internet where a number
of 
people including Patty from Albany and Ray from Rutgers listened in and

participated via text chat.  The call was recorded and is now available

as an MP3 here: 
http://classcaster_news.classcaster.org/blog/resource/podcast/classcaster-20060131-133120.mp3

Overall, I was pleased with how it went and we learned a few things. 
Asterisk and the Classcaster server performed well with no real strain

placed on the server while handling the call, streaming on the net, and

recording.  If you listen to the podcast, you will note that the
quality 
varies and there are echo and feedback issues especially early in the 
call. It seems that in this case quality of equipment matters.  Using a

good quality headset mic definitely makes a difference and using a 
hand-held or table top mic with speakers is not a good idea.

>From a technical point of view the streaming and recording where fine.

  They accurately relayed exactly what we were hearing, the good and 
bad.   The echo problems that are evident in the recording are from end

user equipment, not inherent in the system itself.

The net stream ran cleanly with about a 30-45 second delay that is to
be 
expected as the server transcodes the audio from the GSM codec used by

the SIP phones to the Vorbis codec of the stream and passes if from the

Asterisk PBX to the Icecast streaming server.  This latency can be 
reduced by fine tuning the server. T

he recording was created natively by Asterisk as a mono WAV file with
an 
8K sample rate.  This results in a WAV file that is about 1 megabyte
for 
each minute of recording.  After the call was over I used Audacity to 
normalize the audio and encode it as MP3.

I'm pretty sure we will be doing this again and developing CALI
services 
that make use of this technology.

Thanks,

-- 
Elmer R. Masters
Director of Internet Development
Center for Computer-Assisted Legal Instruction
404-712-2211    emasters at cali.org
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