[asterisk-commits] mjordan: branch certified-1.8.15 r400118 - /certified/branches/1.8.15/res/

[SVN commits to the Asterisk project]

Author: mjordan
Date: Sat Sep 28 17:36:29 2013
New Revision: 400118

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=400118
Log:
res_rtp_asterisk: Correct erroneous lost packet information in RTCP reports

RTCP's calculation of the number of lost packets in an RTP stream is based on
that stream's sequence number count, the number of received packets, and how
many packets we expect to receive. When the SSRC for an RTP stream changes,
there can - and almost always will be - a large jump in the next packet's
timestamp and sequence number. If we don't reset the number of received
packets, sequence number count, and other metrics used by RTCP, the next RR/SR
report will use the previous SSRC's values to calculate the lost packet count
for the new SSRC - resulting in a very large number of lost packets.

This patch modifies res_rtp_asterisk such that, if it detects a SSRC change, it
will reset the various values used by the RTCP calculations. From the
perspective of RTCP, this appears as a new media stream - which is what it is.

Review: https://reviewboard.asterisk.org/r/2886/

(closes issue AST-1174)
Reported by: Thomas Arimont
........

Merged revisions 400089 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Modified:
    certified/branches/1.8.15/res/res_rtp_asterisk.c

Modified: certified/branches/1.8.15/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/certified/branches/1.8.15/res/res_rtp_asterisk.c?view=diff&rev=400118&r1=400117&r2=400118
==============================================================================
--- certified/branches/1.8.15/res/res_rtp_asterisk.c (original)
+++ certified/branches/1.8.15/res/res_rtp_asterisk.c Sat Sep 28 17:36:29 2013
@@ -1138,7 +1138,7 @@
 	}
 
 	if (!res) {
-		/* 
+		/*
 		 * Not being rescheduled.
 		 */
 		ao2_ref(instance, -1);
@@ -2259,6 +2259,13 @@
 
 		f = ast_frisolate(&srcupdate);
 		AST_LIST_INSERT_TAIL(&frames, f, frame_list);
+
+		rtp->seedrxseqno = 0;
+		rtp->rxcount = 0;
+		rtp->cycles = 0;
+		rtp->lastrxseqno = 0;
+		rtp->rtcp->expected_prior = 0;
+		rtp->rtcp->received_prior = 0;
 	}
 
 	rtp->rxssrc = ssrc;
@@ -2791,14 +2798,14 @@
 	payload = ast_rtp_codecs_payload_lookup(ast_rtp_instance_get_codecs(instance), AST_RTP_CN);
 
 	level = 127 - (level & 0x7f);
-	
+
 	rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
 
 	/* Get a pointer to the header */
 	rtpheader = (unsigned int *)data;
 	rtpheader[0] = htonl((2 << 30) | (1 << 23) | (payload.code << 16) | (rtp->seqno++));
 	rtpheader[1] = htonl(rtp->lastts);
-	rtpheader[2] = htonl(rtp->ssrc); 
+	rtpheader[2] = htonl(rtp->ssrc);
 	data[12] = level;
 
 	res = rtp_sendto(instance, (void *) rtpheader, hdrlen + 1, 0, &remote_address);