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Merged revisions 194208 via svnmerge from

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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: https://origsvn.digium.com/svn/asterisk/trunk@194209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2009-05-13 13:39:10 +00:00
parent b399b5389d
commit 1179ecf165
1 changed files with 66 additions and 15 deletions
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81
res/res_rtp_asterisk.c
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@@ -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;