[asterisk-dev] [Code Review] rtp timestamp to timevalcalculation fix
Nick Lewis
Nick.Lewis at atltelecom.com
Thu Jan 21 05:02:59 CST 2010
>Unless the rtp timestamp does not always reflect the number of samples
incremented,
>I don't see any reason not to use this method.
There may be a problem with G.722
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.
_____________________________________________________________________
This message has been checked for all known viruses by Star Internet delivered through the MessageLabs Virus Control Centre.
_____________________________________________________________________
Disclaimer of Liability
ATL Telecom Ltd shall not be held liable for any improper or incorrect use of the information described and/or contained herein and assumes no responsibility for anyones use of the information. In no event shall ATL Telecom Ltd be liable for any direct, indirect, incidental, special, exemplary, or consequential damages (including, but not limited to, procurement or substitute goods or services; loss of use, data, or profits; or business interruption) however caused and on any theory of liability, whether in contract, strict liability, or tort (including negligence or otherwise) arising in any way out of the use of this system, even if advised of the possibility of such damage.
Registered Office: ATL Telecom Ltd, Fountain Lane, St. Mellons Cardiff, CF3 0FB
Registered in Wales Number 4335781
All goods and services supplied by ATL Telecom Ltd are supplied subject to ATL Telecom Ltd standard terms and conditions, available upon request.
More information about the asterisk-dev
mailing list