[Asterisk-Dev] changing codec during call

[Steve Underwood]

Steve Kann wrote:

> Race Vanderdecken wrote:
>
>>Detecting the the bandwidth constraint constraint might be possible
>>using RTCP. http://www.nwfusion.com/news/tech/2003/1117techupdate.html
>>
>>I have not looked to see if Asterisk is using RTCP, but that would be
>>the correct way to control and detect.
>>  
>>
>
> IAX2 now supports sending all of the parameters that are described in 
> the _extended_ RTCP XR stuff you quote there (the basic RTCP RR does 
> not support all of this).
>
> But I still fail to see how you can determine from this information 
> alone, whether reducing your bandwidth usage by 5kbps or whatever is 
> going to affect the call quality in a positive or negative way.
>
> Certainly, it would be good network policy for us to lower our 
> consumption when we see congestion (like TCP does), but it is not 
> necessarily going to improve the quality of our call.

High packet loss rate, or even high jitter, might give us a clue that a 
lower bit rate would be beneficial. I have no idea how to meaningfully 
determine that a higher bit rate would be harmless. Probing with the 
higher rate every time the loss and jitter are low is quite likely to 
cause far too much juggling of rates.

Regards,
Steve