[teknoids] Outcome: Classcaster teleconference webcast

Seibel, Robert rfs at cwsl.edu
Wed Feb 1 14:43:23 EST 2006

I don't know what that was, but I did not hear the word classcaster for 2 whole minutes, so it could not have been Elmer.
Bob Seibel
California Western School of Law
rfs at cwsl.edu
619 525 1458


From: teknoids-bounces at ruckus.law.cornell.edu on behalf of Patricia Baia
Sent: Wed 2/1/2006 11:29 AM
To: teknoids at ruckus.law.cornell.edu
Subject: Re: [teknoids] Outcome: Classcaster teleconference webcast

Great work Elmer!!!
I do have to say, currently this audio clip seems to be my favorite :)  LOL... What is this conversation?   -Patty  
Patricia L. Baia  M.S.
Instructional Technologist & Ph.D Student
Albany Law School
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: 

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.


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