[asterisk-dev] [Code Review] rtp timestamp to timeval calculation fix

David Vossel dvossel at digium.com
Thu Jan 21 12:55:16 CST 2010


----- "Nick Lewis" <Nick.Lewis at atltelecom.com> wrote:

> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/468/#review1387
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> 
> I would like to see it tested with G.722 before it is shipped as I
> think the problem with RFC1890 may apply to the new method
> 
> RFC 3551 Section 4.5.2 states:
>    Even though the actual sampling rate for G.722 audio is 16,000 Hz,
>    the RTP clock rate for the G722 payload format is 8,000 Hz because
>    that value was erroneously assigned in RFC 1890 and must remain
>    unchanged for backward compatibility.  The octet rate or
> sample-pair
>    rate is 8,000 Hz.
> 
> 
> 
> - Nick
> 
> 

The sampling rate for g722 for this calculation will be 8kHz.   rtp_get_rate() is smart enough to return the 8khz for g722.


David Vossel
Digium, Inc. | Software Developer, Open Source Software
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com & www.asterisk.org
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