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res_rtp_asterisk: RTT miscalculation in RTCP

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When retrieving RTCP stats for PJSIP channels, RTT values are unreliable.
RTT calculation is correct, but the data representation isn't.  RTT is
represented by a 32-bit fixed-point number with the integer part in the
first 16 bits and the fractional part in the last 16 bits.  In order to
get the RTT value, the fractional part is miscalculated, there is an
unnecessary 16 bit shift that causes overflow.  Besides this there is
another mistake, when transforming the integer value to the fixed point
fractional part via bitwise operation, that loses precision.

* RTT fractional part is no longer shifted, avoiding overflow.

* RTT fractional part is transformed to its fixed-point value more
precisely.

* Fixed timeval2ntp() and ntp2timeval() second fraction conversions.

* Fixed NTP timestamp report logging.  The usec was inexplicably
multiplied by 4096.

ASTERISK-26566 #close
Reported by Hector Royo Concepcion

Change-Id: Ie09bdabfee75afb3f1b8ddfd963e5219ada3b96f
This commit is contained in:
gestoip2
2016-11-11 14:16:50 +00:00
committed by Richard Mudgett
parent 038158bf7b
commit d9b24cce0a
2 changed files with 42 additions and 9 deletions
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2
main/rtp_engine.c
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@@ -2493,7 +2493,7 @@ static struct ast_manager_event_blob *rtcp_report_to_ami(struct stasis_message *
if (type == AST_RTP_RTCP_SR) {
ast_str_append(&packet_string, 0, "SentNTP: %lu.%06lu\r\n",
(unsigned long)payload->report->sender_information.ntp_timestamp.tv_sec,
(unsigned long)payload->report->sender_information.ntp_timestamp.tv_usec * 4096);
(unsigned long)payload->report->sender_information.ntp_timestamp.tv_usec);
ast_str_append(&packet_string, 0, "SentRTP: %u\r\n",
payload->report->sender_information.rtp_timestamp);
ast_str_append(&packet_string, 0, "SentPackets: %u\r\n",