diff options
| author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-09 19:52:45 +0000 |
|---|---|---|
| committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-12-09 19:52:45 +0000 |
| commit | a6984bf329c715d85fc04db36f814cc355e8dc21 (patch) | |
| tree | 18c552f5b6ea0bb397aaa93dce8b1942e72886d7 /main/rtp.c | |
| parent | 115f48007cfd9eec24422e465454c433c514ba24 (diff) | |
Merged revisions 162205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk
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r162205 | file | 2008-12-09 15:48:35 -0400 (Tue, 09 Dec 2008) | 14 lines
Merged revisions 162204 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r162204 | file | 2008-12-09 15:47:07 -0400 (Tue, 09 Dec 2008) | 7 lines
Make sure that the timestamp for DTMF is not the same as the previous voice frame and do not send audio when transmitting DTMF as this confuses some equipment.
(closes issue #13209)
Reported by: ip-rob
Patches:
13209.diff uploaded by file (license 11)
Tested by: ip-rob, bujones
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git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@162207 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'main/rtp.c')
| -rw-r--r-- | main/rtp.c | 8 |
1 files changed, 6 insertions, 2 deletions
diff --git a/main/rtp.c b/main/rtp.c index 6af9b4886..a98ed6858 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -3050,6 +3050,7 @@ int ast_rtp_senddigit_begin(struct ast_rtp *rtp, char digit) rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000)); rtp->send_duration = 160; + rtp->lastdigitts = rtp->lastts + rtp->send_duration; /* Get a pointer to the header */ rtpheader = (unsigned int *)data; @@ -3169,10 +3170,9 @@ int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit) ast_inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), rtp->send_payload, rtp->seqno, rtp->lastdigitts, res - hdrlen); } + rtp->lastts += rtp->send_duration; rtp->sending_digit = 0; rtp->send_digit = 0; - /* Increment lastdigitts */ - rtp->lastdigitts += 960; rtp->seqno++; return res; @@ -3498,6 +3498,10 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec int pred; int mark = 0; + if (rtp->sending_digit) { + return 0; + } + ms = calc_txstamp(rtp, &f->delivery); /* Default prediction */ if (f->frametype == AST_FRAME_VOICE) { |