Correct output of RTCP jitter statistics in SR and RR reports
authorKinsey Moore <kmoore@digium.com>
Thu, 19 Jan 2012 22:44:38 +0000 (22:44 +0000)
committerKinsey Moore <kmoore@digium.com>
Thu, 19 Jan 2012 22:44:38 +0000 (22:44 +0000)
commitadd6efc20cec066e753879a552f01a5e2a6058fa
tree79bdaf8a31bedf4e4cccabccf95bf3cd3e087685
parent6fd0ac9dcd8225b060c0038ee0cd5440040a57ef
Correct output of RTCP jitter statistics in SR and RR reports

Change the RTCP RR and SR generation code to convert Asterisk's internal jitter
statistics to be represented in RTP timestamp units based on the rate of the
codec in use instead of in seconds.

(closes issue ASTERISK-14530)
........

Merged revisions 351611 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 351612 from http://svn.asterisk.org/svn/asterisk/branches/10

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res/res_rtp_asterisk.c