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Merge "res_rtp_asterisk: Fix sequence number cycling and packet loss count."

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This commit is contained in:
Friendly Automation
2019-05-15 17:45:56 -05:00
committed by Gerrit Code Review
parent e222dc71d5 7a6fd83aca
commit 0575a8923b
1 changed files with 9 additions and 1 deletions
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10
res/res_rtp_asterisk.c
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@@ -4126,6 +4126,14 @@ static void calculate_lost_packet_statistics(struct ast_rtp *rtp,
*lost_packets = expected_packets - rtp->rxcount;
expected_interval = expected_packets - rtp->rtcp->expected_prior;
received_interval = rtp->rxcount - rtp->rtcp->received_prior;
if (received_interval > expected_interval) {
/* If we receive some late packets it is possible for the packets
* we received in this interval to exceed the number we expected.
* We update the expected so that the packet loss calculations
* show that no packets are lost.
*/
expected_interval = received_interval;
}
lost_interval = expected_interval - received_interval;
if (expected_interval == 0 || lost_interval <= 0) {
*fraction_lost = 0;
@@ -6801,7 +6809,7 @@ static struct ast_frame *ast_rtp_interpret(struct ast_rtp_instance *instance, st
ast_log(LOG_WARNING, "scheduling RTCP transmission failed.\n");
}
}
if ((int)rtp->lastrxseqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */
if ((int)prev_seqno - (int)seqno > 100) /* if so it would indicate that the sender cycled; allow for misordering */
rtp->cycles += RTP_SEQ_MOD;
/* If we are directly bridged to another instance send the audio directly out,