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res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets.

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With this change, the initial RTP sequence number is randomly chosen not between
0 and 65535 (0xffff) but 0 and 32767 (0x7fff). This assures, the roll-over
counter (ROC) synchronization is not lost for sRTP, when the very first RTP
packets get lost; see http://srtp.sourceforge.net/faq.html#Q6

ASTERISK-26207 #close

Change-Id: I9a527e3aa3ce8f3becc5131d7ba32b57b5845464
This commit is contained in:
Alexander Traud
2016-07-18 12:13:25 +02:00
parent 26b4760808
commit cb5e3445be
1 changed files with 1 additions and 1 deletions
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res/res_rtp_asterisk.c
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@@ -2592,7 +2592,7 @@ static int ast_rtp_new(struct ast_rtp_instance *instance,
/* Set default parameters on the newly created RTP structure */
rtp->ssrc = ast_random();
rtp->seqno = ast_random() & 0xffff;
rtp->seqno = ast_random() & 0x7fff;
rtp->strict_rtp_state = (strictrtp ? STRICT_RTP_LEARN : STRICT_RTP_OPEN);
if (strictrtp) {
rtp_learning_seq_init(&rtp->rtp_source_learn, (uint16_t)rtp->seqno);