[Asterisk-Dev] changing codec during call

[Michael Giagnocavo]

Not knowing much about this at all, I ask why wouldn't the jitterbuffer
handle this monitoring, as it has the most details of what's going on (such
as packet loss, where switching to a packet-loss-friendly codec would be a
good idea).

-Michael

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Jesse Kaijen
Sent: Friday, February 25, 2005 8:40 AM
To: asterisk-dev at lists.digium.com
Subject: [Asterisk-Dev] changing codec during call

Hello I'm a student and for my bachelor-assignment I'm looking into VoIP.
I'm researching if the audio perceptive of the end-user will get higher if
during a call a switch of codec is made.
I was wondering if it's possible to switch codec's during a call with IAX.
Can someone help me on that?

This is the idea:
During a call with ulaw (64kb) the available bandwidth drops from 80kb
(sufficient) to 40kb for a longer period. The losses are great and the
call-quality is horrible. At that point changing codec to GSM for instance
may result in a better quality. When the bandwidth is restored change the
codec back. A monitor must listen after a jitterbuffer and then decide to
change codec. 

Picture:
                        +----*asterisk*-----+
UA--->---->|---->up---->|--jitbuf---decoder-|--PSTN
UA---jitbuf|<---down<---|-----------encoder-|--PSTN
   ^                    +-------------------+
   |
 point where the monitor must listen

My question is (if possible) which command do I have to send during a call
to switch codecs? And can the current iax-clients handle a codec change?

Greetings

Jesse Kaijen
jesse at kayen.nl


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