diff options
author | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-05-13 13:39:10 +0000 |
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committer | file <file@f38db490-d61c-443f-a65b-d21fe96a405b> | 2009-05-13 13:39:10 +0000 |
commit | 26a4949b2d61b9455cf5b3cefff9c0309ab73990 (patch) | |
tree | f934ce47c3190565b066687922d6371d06450db7 /res/res_rtp_asterisk.c | |
parent | 8191b1aea14fa4cd2d47b561633105ea9e08556e (diff) |
Merged revisions 194208 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) | 11 lines
Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over.
(closes issue #14815)
Reported by: geoff2010
Patches:
v1-14815.patch uploaded by dimas (license 88)
Tested by: geoff2010, file, dimas, ZX81, moliveras
(closes issue #14460)
Reported by: moliveras
Tested by: moliveras
........
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@194209 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'res/res_rtp_asterisk.c')
-rw-r--r-- | res/res_rtp_asterisk.c | 81 |
1 files changed, 66 insertions, 15 deletions
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c index 4bd5e8895..a6a4f3caf 100644 --- a/res/res_rtp_asterisk.c +++ b/res/res_rtp_asterisk.c @@ -72,7 +72,7 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$") #define RTP_MTU 1200 -#define DEFAULT_DTMF_TIMEOUT 3000 /*!< samples */ +#define DEFAULT_DTMF_TIMEOUT (150 * (8000 / 1000)) /*!< samples */ #define ZFONE_PROFILE_ID 0x505a @@ -139,7 +139,8 @@ struct ast_rtp { /* DTMF Reception Variables */ char resp; unsigned int lastevent; - int dtmfcount; + unsigned int dtmf_duration; /*!< Total duration in samples since the digit start event */ + unsigned int dtmf_timeout; /*!< When this timestamp is reached we consider END frame lost and forcibly abort digit */ unsigned int dtmfsamples; /* DTMF Transmission Variables */ unsigned int lastdigitts; @@ -1335,23 +1336,59 @@ static struct ast_frame *process_dtmf_rfc2833(struct ast_rtp_instance *instance, if (ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE)) { if ((rtp->lastevent != timestamp) || (rtp->resp && rtp->resp != resp)) { rtp->resp = resp; - rtp->dtmfcount = 0; + rtp->dtmf_timeout = 0; f = send_dtmf(instance, AST_FRAME_DTMF_END, ast_rtp_instance_get_prop(instance, AST_RTP_PROPERTY_DTMF_COMPENSATE)); f->len = 0; rtp->lastevent = timestamp; } } else { - if ((!(rtp->resp) && (!(event_end & 0x80))) || (rtp->resp && rtp->resp != resp)) { - rtp->resp = resp; - f = send_dtmf(instance, AST_FRAME_DTMF_BEGIN, 0); - rtp->dtmfcount = dtmftimeout; - } else if ((event_end & 0x80) && (rtp->lastevent != seqno) && rtp->resp) { - f = send_dtmf(instance, AST_FRAME_DTMF_END, 0); - f->len = ast_tvdiff_ms(ast_samp2tv(samples, 8000), ast_tv(0, 0)); /* XXX hard coded 8kHz */ - rtp->resp = 0; - rtp->dtmfcount = 0; - rtp->lastevent = seqno; + /* The duration parameter measures the complete + duration of the event (from the beginning) - RFC2833. + Account for the fact that duration is only 16 bits long + (about 8 seconds at 8000 Hz) and can wrap is digit + is hold for too long. */ + unsigned int new_duration = rtp->dtmf_duration; + unsigned int last_duration = new_duration & 0xFFFF; + + if (last_duration > 64000 && samples < last_duration) { + new_duration += 0xFFFF + 1; + } + new_duration = (new_duration & ~0xFFFF) | samples; + + if (event_end & 0x80) { + /* End event */ + if ((rtp->lastevent != seqno) && rtp->resp) { + rtp->dtmf_duration = new_duration; + f = send_dtmf(instance, AST_FRAME_DTMF_END, 0); + f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, 8000), ast_tv(0, 0)); + rtp->resp = 0; + rtp->dtmf_duration = rtp->dtmf_timeout = 0; + } + } else { + /* Begin/continuation */ + + if (rtp->resp && rtp->resp != resp) { + /* Another digit already began. End it */ + f = send_dtmf(instance, AST_FRAME_DTMF_END, 0); + f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, 8000), ast_tv(0, 0)); + rtp->resp = 0; + rtp->dtmf_duration = rtp->dtmf_timeout = 0; + } + + if (rtp->resp) { + /* Digit continues */ + rtp->dtmf_duration = new_duration; + } else { + /* New digit began */ + rtp->resp = resp; + f = send_dtmf(instance, AST_FRAME_DTMF_BEGIN, 0); + rtp->dtmf_duration = samples; + } + + rtp->dtmf_timeout = timestamp + rtp->dtmf_duration + dtmftimeout; } + + rtp->lastevent = seqno; } rtp->dtmfsamples = samples; @@ -1432,7 +1469,7 @@ static struct ast_frame *process_dtmf_cisco(struct ast_rtp_instance *instance, u rtp->resp = 0; } else if (rtp->resp == resp) rtp->dtmfsamples += 20 * 8; - rtp->dtmfcount = dtmftimeout; + rtp->dtmf_timeout = 0; return f; } @@ -1982,6 +2019,20 @@ static struct ast_frame *ast_rtp_read(struct ast_rtp_instance *instance, int rtc rtp->f.frametype = (rtp->f.subclass & AST_FORMAT_AUDIO_MASK) ? AST_FRAME_VOICE : (rtp->f.subclass & AST_FORMAT_VIDEO_MASK) ? AST_FRAME_VIDEO : AST_FRAME_TEXT; rtp->rxseqno = seqno; + + if (rtp->dtmf_timeout && rtp->dtmf_timeout < timestamp) { + rtp->dtmf_timeout = 0; + + if (rtp->resp) { + struct ast_frame *f; + f = send_dtmf(instance, AST_FRAME_DTMF_END, 0); + f->len = ast_tvdiff_ms(ast_samp2tv(rtp->dtmf_duration, 8000), ast_tv(0, 0)); + rtp->resp = 0; + rtp->dtmf_timeout = rtp->dtmf_duration = 0; + return f; + } + } + rtp->lastrxts = timestamp; rtp->f.src = "RTP"; @@ -2522,7 +2573,7 @@ static int rtp_reload(int reload) } if ((s = ast_variable_retrieve(cfg, "general", "dtmftimeout"))) { dtmftimeout = atoi(s); - if ((dtmftimeout < 0) || (dtmftimeout > 20000)) { + if ((dtmftimeout < 0) || (dtmftimeout > 64000)) { ast_log(LOG_WARNING, "DTMF timeout of '%d' outside range, using default of '%d' instead\n", dtmftimeout, DEFAULT_DTMF_TIMEOUT); dtmftimeout = DEFAULT_DTMF_TIMEOUT; |