]> 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 14:29:25 +0000 (09:29 -0500)
committerGeorge Joseph <gjoseph@digium.com>
Tue, 11 Dec 2018 15:06:02 +0000 (10:06 -0500)
This reverts commit cf620ce0f6dc4a7ef713dc85b4c2610b614cd647.

Pending resolution of ASTERISK_28200

Change-Id: If18ab1166db0d634ce8a099bd7460251e9c682e1

res/res_rtp_asterisk.c

index 4586ca67f6c7cef930612a571b614c8cb563e665..8841f6d6de935829aeaec8f527ed573a99cf8990 100644 (file)
@@ -5732,16 +5732,6 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc
        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;