Change RFC2833 DTMF event duration on end to report actual elapsed time.

The scenario here is with a non P2P early media session. The reported time
length of DTMF presses are coming up short when sending to the remote side.
Currently the event duration is a running total that is incremented when sending
continuation packets. These continuation packets are only triggered upon
incoming media from the remote side, which means that the running total probably
is not going to end up matching the actual length of time Asterisk received
DTMF. This patch changes the end event duration to be lengthened if it is
detected that the end event is going to come up short.

Review: https://reviewboard.asterisk.org/r/957/

ABE-2476

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@289797 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 843fa2dd6c960d18b9db1afe4cdcc0fd97e7e77f..109b46e83aa64fd96f2dd30824bf009aba96229a 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4182,7 +4182,7 @@ static int sip_senddigit_end(struct ast_channel *ast, char digit, unsigned int d
                break;
        case SIP_DTMF_RFC2833:
                if (p->rtp)
-                       ast_rtp_senddigit_end(p->rtp, digit);
+                       ast_rtp_senddigit_end_with_duration(p->rtp, digit, duration);
                break;
        case SIP_DTMF_INBAND:
                res = -1; /* Tell Asterisk to stop inband indications */

diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index 14e4b83967519e8c4e0e246184d87b58d99622fa..8c54ece6d48531d6392eacc7a63dccf95defb2ff 100644 (file)
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -174,6 +174,7 @@ int ast_rtcp_fd(struct ast_rtp *rtp);
 int ast_rtp_senddigit_begin(struct ast_rtp *rtp, char digit);
 
 int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit);
+int ast_rtp_senddigit_end_with_duration(struct ast_rtp *rtp, char digit, unsigned int duration);
 
 int ast_rtp_sendcng(struct ast_rtp *rtp, int level);
 

diff --git a/main/rtp.c b/main/rtp.c
index 5a1266a3c8ce06dfcfadc1251be9c0be147e7d82..3133de7d9038a4af6bd448d42bfae18f25c7349d 100644 (file)
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -186,6 +186,7 @@ static int ast_rtcp_write_rr(const void *data);
 static unsigned int ast_rtcp_calc_interval(struct ast_rtp *rtp);
 static int ast_rtp_senddigit_continuation(struct ast_rtp *rtp);
 int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit);
+int ast_rtp_senddigit_end_with_duration(struct ast_rtp *rtp, char digit, unsigned int duration);
 
 #define FLAG_3389_WARNING              (1 << 0)
 #define FLAG_NAT_ACTIVE                        (3 << 1)
@@ -2421,12 +2422,18 @@ static int ast_rtp_senddigit_continuation(struct ast_rtp *rtp)
        return 0;
 }
 
-/*! \brief Send end packets for DTMF */
 int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit)
+{
+       return ast_rtp_senddigit_end_with_duration(rtp, digit, 0);
+}
+
+/*! \brief Send end packets for DTMF */
+int ast_rtp_senddigit_end_with_duration(struct ast_rtp *rtp, char digit, unsigned int duration)
 {
        unsigned int *rtpheader;
        int hdrlen = 12, res = 0, i = 0;
        char data[256];
+       unsigned int measured_samples;
        
        /* If no address, then bail out */
        if (!rtp->them.sin_addr.s_addr || !rtp->them.sin_port)
@@ -2449,6 +2456,13 @@ int ast_rtp_senddigit_end(struct ast_rtp *rtp, char digit)
 
        rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
 
+       if (duration > 0 && (measured_samples = duration * rtp_get_rate(rtp->f.subclass) / 1000) > rtp->send_duration) {
+               if (option_debug > 1) {
+                       ast_log(LOG_DEBUG, "Adjusting final end duration from %u to %u\n", rtp->send_duration, measured_samples);
+               }
+               rtp->send_duration = measured_samples;
+       }
+
        rtpheader = (unsigned int *)data;
        rtpheader[1] = htonl(rtp->lastdigitts);
        rtpheader[2] = htonl(rtp->ssrc);