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

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@400089 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index 6fb919504058c6c80a5c3f08d57605245de455c5..50cbfc8d944debef8a2d8f631f3b2a14e1401199 100644 (file)
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -2275,6 +2275,12 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
                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->last_seqno = 0;
                rtp->last_end_timestamp = 0;
        }