Calculate the timestamp for outbound RTP if we don't have timing information

This patch calculates the timestamp for outbound RTP when we don't have timing
information. This uses the same approach in res_rtp_asterisk. Thanks to both
Pietro and Tzafrir for providing patches.

(closes issue ASTERISK-19883)
Reported by: Giacomo Trovato
Tested by: Pietro Bertera, Tzafrir Cohen
patches:
  rtp-timestamp-1.8.patch uploaded by tzafrir (License 5035)
  rtp-timestamp.patch uploaded by pbertera (License 5943)

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385636 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/res/res_rtp_multicast.c b/res/res_rtp_multicast.c
index d5b1c143d70cd980e3c04a46c700be899b40eae7..2b5cbbc82aaaac3e84d4e6ec24841a696d05fd5b 100644 (file)
--- a/res/res_rtp_multicast.c
+++ b/res/res_rtp_multicast.c
@@ -90,6 +90,8 @@ struct multicast_rtp {
        unsigned int ssrc;
        /*! Sequence number, used when creating/sending the RTP packet */
        uint16_t seqno;
+       unsigned int lastts;
+       struct timeval txcore;
 };
 
 /* Forward Declarations */
@@ -140,6 +142,30 @@ static int multicast_rtp_new(struct ast_rtp_instance *instance, struct sched_con
        return 0;
 }
 
+static int rtp_get_rate(format_t format)
+{
+        return (format == AST_FORMAT_G722) ? 8000 : ast_format_rate(format);
+}
+
+static unsigned int calc_txstamp(struct multicast_rtp *rtp, struct timeval *delivery)
+{
+        struct timeval t;
+        long ms;
+
+        if (ast_tvzero(rtp->txcore)) {
+                rtp->txcore = ast_tvnow();
+                rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000;
+        }
+
+        t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
+        if ((ms = ast_tvdiff_ms(t, rtp->txcore)) < 0) {
+                ms = 0;
+        }
+        rtp->txcore = t;
+
+        return (unsigned int) ms;
+}
+
 /*! \brief Helper function which populates a control packet with useful information and sends it */
 static int multicast_send_control_packet(struct ast_rtp_instance *instance, struct multicast_rtp *multicast, int command)
 {
@@ -209,12 +235,15 @@ static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_fra
        struct ast_frame *f = frame;
        struct ast_sockaddr remote_address;
        int hdrlen = 12, res = 0, codec;
+       int rate;
        unsigned char *rtpheader;
+       unsigned int ms = calc_txstamp(multicast, &frame->delivery);
 
        /* We only accept audio, nothing else */
        if (frame->frametype != AST_FRAME_VOICE) {
                return 0;
        }
+       rate = rtp_get_rate(frame->subclass.codec) / 1000;
 
        /* Grab the actual payload number for when we create the RTP packet */
        if ((codec = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(instance), 1, frame->subclass.codec)) < 0) {
@@ -226,11 +255,20 @@ static int multicast_rtp_write(struct ast_rtp_instance *instance, struct ast_fra
                f = ast_frdup(frame);
        }
 
+       /* Calucate last TS */
+       multicast->lastts = multicast->lastts + ms * rate;
+
        /* Construct an RTP header for our packet */
        rtpheader = (unsigned char *)(f->data.ptr - hdrlen);
        put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (multicast->seqno)));
-       put_unaligned_uint32(rtpheader + 4, htonl(f->ts * 8));
-       put_unaligned_uint32(rtpheader + 8, htonl(multicast->ssrc));
+       put_unaligned_uint32(rtpheader + 4, htonl(multicast->lastts));
+
+       if (ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO)) {
+               put_unaligned_uint32(rtpheader + 4, htonl(f->ts * 8));
+       }
+       else {
+               put_unaligned_uint32(rtpheader + 8, htonl(multicast->ssrc));
+       }
 
        /* Increment sequence number and wrap to 0 if it overflows 16 bits. */
        multicast->seqno = 0xFFFF & (multicast->seqno + 1);