[cisco-voip] g711 sample size in CCM - 10ms or 20ms

[Wes Sisk]

Yes, codec is somewhat independent of sample size.  The two are 
typically related by available buffer memory in the endpoint.  The 
receiving endpoint must implement a de-jitter buffer, otherwise voice 
quality will directly suffer.  de-jitter buffer is simply a space of 
memory to receive and sort RTP payload before playout.  high bit rate 
codecs such as g711 take more space to implement a 40 msec dejitter 
buffer compared to low bit rate codecs such as g729.

AFAIK all protocols advertise RECIEVE capabilities, not transmit 
capabilities.  This is a subtle difference as the two should basically 
match.

I know in CCM 3.0 and 3.1 it was possible to setup asymetric audio 
streams so

ep1                                ep2
g711(20)------------->
              <-------------g729(20)

I used to see this when a 12SP+ or 30VIP called a 7960.  I have not seen 
this in a long time.  The call actually worked fine.

/Wes

Eric Knudson wrote:

>additionally, just for my edification - isn't codec independent of
>sample size? Additionally, isn't each rtp stream independent from
>another - for example, you could have:
>
>endpoint                  endpoint
>g.711(10ms)----------->
>           <--------------g.711(160ms)
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