]> git.ipfire.org Git - thirdparty/asterisk.git/commitdiff
Revert "RTP: reset DTMF last seqno/timestamp on voice packet with marker bit"
authorGeorge Joseph <gjoseph@digium.com>
Tue, 11 Dec 2018 14:28:18 +0000 (09:28 -0500)
committerGeorge Joseph <gjoseph@digium.com>
Tue, 11 Dec 2018 14:28:18 +0000 (09:28 -0500)
This reverts commit 331c906c4811df17612efa5c31e19df7186b1c81.

Pending resolution of ASTERISK_28200

Change-Id: Ie7172707b603c1da3f200613bd4473335af75128

res/res_rtp_asterisk.c

index ff8e453055a774f2a0b5d18d9e444b9993b7629e..465f9bc23ac51e381bf88d74feb00d1fb700243c 100644 (file)
@@ -6438,16 +6438,6 @@ static struct ast_frame *ast_rtp_interpret(struct ast_rtp_instance *instance, st
        switch (ast_format_get_type(rtp->f.subclass.format)) {
        case AST_MEDIA_TYPE_AUDIO:
                rtp->f.frametype = AST_FRAME_VOICE;
-
-               /* The marker bit set on the voice packet indicates the start
-                * of a new stream and a new time stamp. Need to reset the DTMF
-                * last sequence number and the timestamp of the last END packet.
-                */
-               if (mark) {
-                       rtp->last_seqno = 0;
-                       rtp->last_end_timestamp = 0;
-               }
-
                break;
        case AST_MEDIA_TYPE_VIDEO:
                rtp->f.frametype = AST_FRAME_VIDEO;