res_rtp_asterisk: RTT miscalculation in RTCP

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

main/rtp_engine.c

@@ -1987,7 +1987,7 @@ 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",

res/res_rtp_asterisk.c

@@ -3086,7 +3086,26 @@ static void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw)
 	unsigned int sec, usec, frac;
 	sec = tv.tv_sec + 2208988800u; /* Sec between 1900 and 1970 */
 	usec = tv.tv_usec;
-	frac = (usec << 12) + (usec << 8) - ((usec * 3650) >> 6);
+	/*
+	 * Convert usec to 0.32 bit fixed point without overflow.
+	 *
+	 * = usec * 2^32 / 10^6
+	 * = usec * 2^32 / (2^6 * 5^6)
+	 * = usec * 2^26 / 5^6
+	 *
+	 * The usec value needs 20 bits to represent 999999 usec.  So
+	 * splitting the 2^26 to get the most precision using 32 bit
+	 * values gives:
+	 *
+	 * = ((usec * 2^12) / 5^6) * 2^14
+	 *
+	 * Splitting the division into two stages preserves all the
+	 * available significant bits of usec over doing the division
+	 * all at once.
+	 *
+	 * = ((((usec * 2^12) / 5^3) * 2^7) / 5^3) * 2^7
+	 */
+	frac = ((((usec << 12) / 125) << 7) / 125) << 7;
 	*msw = sec;
 	*lsw = frac;
 }
@@ -3094,7 +3113,8 @@ static void timeval2ntp(struct timeval tv, unsigned int *msw, unsigned int *lsw)
 static void ntp2timeval(unsigned int msw, unsigned int lsw, struct timeval *tv)
 {
 	tv->tv_sec = msw - 2208988800u;
-	tv->tv_usec = ((lsw << 6) / 3650) - (lsw >> 12) - (lsw >> 8);
+	/* Reverse the sequence in timeval2ntp() */
+	tv->tv_usec = ((((lsw >> 7) * 125) >> 7) * 125) >> 12;
 }
 
 static void calculate_lost_packet_statistics(struct ast_rtp *rtp,
@@ -3279,9 +3299,9 @@ static int ast_rtcp_write_report(struct ast_rtp_instance *instance, int sr)
 				ast_sockaddr_stringify(&remote_address), ice ? " (via ICE)" : "");
 		ast_verbose("  Our SSRC: %u\n", rtcp_report->ssrc);
 		if (sr) {
-			ast_verbose("  Sent(NTP): %u.%010u\n",
+			ast_verbose("  Sent(NTP): %u.%06u\n",
 				(unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_sec,
-				(unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_usec * 4096);
+				(unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_usec);
 			ast_verbose("  Sent(RTP): %u\n", rtcp_report->sender_information.rtp_timestamp);
 			ast_verbose("  Sent packets: %u\n", rtcp_report->sender_information.packet_count);
 			ast_verbose("  Sent octets: %u\n", rtcp_report->sender_information.octet_count);
@@ -4020,9 +4040,22 @@ static int update_rtt_stats(struct ast_rtp *rtp, unsigned int lsr, unsigned int
 	lsr_a = ((msw & 0x0000ffff) << 16) | ((lsw & 0xffff0000) >> 16);
 	rtt = lsr_a - lsr - dlsr;
 	rtt_msw = (rtt & 0xffff0000) >> 16;
-	rtt_lsw = (rtt & 0x0000ffff) << 16;
+	rtt_lsw = (rtt & 0x0000ffff);
 	rtt_tv.tv_sec = rtt_msw;
-	rtt_tv.tv_usec = ((rtt_lsw << 6) / 3650) - (rtt_lsw >> 12) - (rtt_lsw >> 8);
+	/*
+	 * Convert 16.16 fixed point rtt_lsw to usec without
+	 * overflow.
+	 *
+	 * = rtt_lsw * 10^6 / 2^16
+	 * = rtt_lsw * (2^6 * 5^6) / 2^16
+	 * = rtt_lsw * 5^6 / 2^10
+	 *
+	 * The rtt_lsw value is in 16.16 fixed point format and 5^6
+	 * requires 14 bits to represent.  We have enough space to
+	 * directly do the conversion because there is no integer
+	 * component in rtt_lsw.
+	 */
+	rtt_tv.tv_usec = (rtt_lsw * 15625) >> 10;
 	rtp->rtcp->rtt = (double)rtt_tv.tv_sec + ((double)rtt_tv.tv_usec / 1000000);
 	if (lsr_a - dlsr < lsr) {
 		return 1;
@@ -4218,9 +4251,9 @@ static struct ast_frame *ast_rtcp_read(struct ast_rtp_instance *instance)
 					&rtcp_report->sender_information.ntp_timestamp);
 			rtcp_report->sender_information.rtp_timestamp = ntohl(rtcpheader[i + 2]);
 			if (rtcp_debug_test_addr(&addr)) {
-				ast_verbose("NTP timestamp: %u.%010u\n",
+				ast_verbose("NTP timestamp: %u.%06u\n",
 						(unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_sec,
-						(unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_usec * 4096);
+						(unsigned int)rtcp_report->sender_information.ntp_timestamp.tv_usec);
 				ast_verbose("RTP timestamp: %u\n", rtcp_report->sender_information.rtp_timestamp);
 				ast_verbose("SPC: %u\tSOC: %u\n",
 						rtcp_report->sender_information.packet_count,