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authormarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>2003-02-05 19:26:49 +0000
committermarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>2003-02-05 19:26:49 +0000
commit0c1a35f2f940ae7d1dbb2b198fd933c94cbaae4b (patch)
treeb8c0bc170731fe5c5311e0a4cad802d6bf6ec20e /rtp.c
parent4c19328ff6ea16ea45068961c50287dc0258834e (diff)
Version 0.3.0 from FTP
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@603 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'rtp.c')
-rwxr-xr-xrtp.c170
1 files changed, 161 insertions, 9 deletions
diff --git a/rtp.c b/rtp.c
index 5162193f1..2a015b6e4 100755
--- a/rtp.c
+++ b/rtp.c
@@ -31,6 +31,11 @@
#include <asterisk/options.h>
#include <asterisk/channel.h>
+#define TYPE_SILENCE 0x2
+#define TYPE_HIGH 0x0
+#define TYPE_LOW 0x1
+#define TYPE_MASK 0x3
+
static int dtmftimeout = 300; /* 300 samples */
struct ast_rtp {
@@ -41,6 +46,7 @@ struct ast_rtp {
unsigned int ssrc;
unsigned int lastts;
unsigned int lastrxts;
+ int lasttxformat;
int dtmfcount;
struct sockaddr_in us;
struct sockaddr_in them;
@@ -56,6 +62,40 @@ struct ast_rtp {
};
+static int g723_len(unsigned char buf)
+{
+ switch(buf & TYPE_MASK) {
+ case TYPE_MASK:
+ case TYPE_SILENCE:
+ return 4;
+ break;
+ case TYPE_HIGH:
+ return 24;
+ break;
+ case TYPE_LOW:
+ return 20;
+ break;
+ default:
+ ast_log(LOG_WARNING, "Badly encoded frame (%d)\n", buf & TYPE_MASK);
+ }
+ return -1;
+}
+
+static int g723_samples(unsigned char *buf, int maxlen)
+{
+ int pos = 0;
+ int samples = 0;
+ int res;
+ while(pos < maxlen) {
+ res = g723_len(buf[pos]);
+ if (res < 0)
+ break;
+ samples += 240;
+ pos += res;
+ }
+ return samples;
+}
+
void ast_rtp_set_data(struct ast_rtp *rtp, void *data)
{
rtp->data = data;
@@ -72,7 +112,7 @@ static void send_dtmf(struct ast_rtp *rtp)
rtp->f.frametype = AST_FRAME_DTMF;
rtp->f.subclass = rtp->resp;
rtp->f.datalen = 0;
- rtp->f.timelen = 0;
+ rtp->f.samples = 0;
rtp->f.mallocd = 0;
rtp->f.src = "RTP";
rtp->resp = 0;
@@ -185,6 +225,9 @@ static int rtpread(int *id, int fd, short events, void *cbdata)
} else if (payloadtype == 121) {
/* CISCO proprietary DTMF bridge */
process_type121(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
+ } else if (payloadtype == 100) {
+ /* CISCO's notso proprietary DTMF bridge */
+ process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
} else {
ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
}
@@ -222,22 +265,25 @@ static int rtpread(int *id, int fd, short events, void *cbdata)
switch(rtp->f.subclass) {
case AST_FORMAT_ULAW:
case AST_FORMAT_ALAW:
- rtp->f.timelen = rtp->f.datalen / 8;
+ rtp->f.samples = rtp->f.datalen;
break;
case AST_FORMAT_SLINEAR:
- rtp->f.timelen = rtp->f.datalen / 16;
+ rtp->f.samples = rtp->f.datalen / 2;
break;
case AST_FORMAT_GSM:
- rtp->f.timelen = 20 * (rtp->f.datalen / 33);
+ rtp->f.samples = 160 * (rtp->f.datalen / 33);
break;
case AST_FORMAT_ADPCM:
- rtp->f.timelen = rtp->f.datalen / 4;
+ rtp->f.samples = rtp->f.datalen * 2;
break;
case AST_FORMAT_G729A:
- rtp->f.timelen = rtp->f.datalen;
+ rtp->f.samples = rtp->f.datalen * 8;
+ break;
+ case AST_FORMAT_G723_1:
+ rtp->f.samples = g723_samples(rtp->f.data, rtp->f.datalen);
break;
default:
- ast_log(LOG_NOTICE, "Unable to calculate timelen for format %d\n", rtp->f.subclass);
+ ast_log(LOG_NOTICE, "Unable to calculate samples for format %d\n", rtp->f.subclass);
break;
}
rtp->f.src = "RTP";
@@ -330,12 +376,27 @@ struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io)
return rtp;
}
+int ast_rtp_settos(struct ast_rtp *rtp, int tos)
+{
+ int res;
+ if ((res = setsockopt(rtp->s, SOL_IP, IP_TOS, &tos, sizeof(tos))))
+ ast_log(LOG_WARNING, "Unable to set TOS to %d\n", tos);
+ return res;
+}
+
void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
{
rtp->them.sin_port = them->sin_port;
rtp->them.sin_addr = them->sin_addr;
}
+void ast_rtp_get_peer(struct ast_rtp *rtp, struct sockaddr_in *them)
+{
+ them->sin_family = AF_INET;
+ them->sin_port = rtp->them.sin_port;
+ them->sin_addr = rtp->them.sin_addr;
+}
+
void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us)
{
memcpy(us, &rtp->us, sizeof(rtp->us));
@@ -365,6 +426,67 @@ static unsigned int calc_txstamp(struct ast_rtp *rtp)
return ms;
}
+int ast_rtp_senddigit(struct ast_rtp *rtp, char digit)
+{
+ unsigned int *rtpheader;
+ int hdrlen = 12;
+ int res;
+ int ms;
+ int pred;
+ int x;
+ char data[256];
+
+ if ((digit <= '9') && (digit >= '0'))
+ digit -= '0';
+ else if (digit == '*')
+ digit = 10;
+ else if (digit == '#')
+ digit = 11;
+ else if ((digit >= 'A') && (digit <= 'D'))
+ digit = digit - 'A' + 12;
+ else if ((digit >= 'a') && (digit <= 'd'))
+ digit = digit - 'a' + 12;
+ else {
+ ast_log(LOG_WARNING, "Don't know how to represent '%c'\n", digit);
+ return -1;
+ }
+
+
+ /* If we have no peer, return immediately */
+ if (!rtp->them.sin_addr.s_addr)
+ return 0;
+
+ ms = calc_txstamp(rtp);
+ /* Default prediction */
+ pred = ms * 8;
+
+ /* Get a pointer to the header */
+ rtpheader = (unsigned int *)data;
+ rtpheader[0] = htonl((2 << 30) | (1 << 23) | (101 << 16) | (rtp->seqno++));
+ rtpheader[1] = htonl(rtp->lastts);
+ rtpheader[2] = htonl(rtp->ssrc);
+ rtpheader[3] = htonl((digit << 24) | (0xa << 16) | (0));
+ for (x=0;x<4;x++) {
+ if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
+ res = sendto(rtp->s, (void *)rtpheader, hdrlen + 4, 0, &rtp->them, sizeof(rtp->them));
+ if (res <0)
+ ast_log(LOG_NOTICE, "RTP Transmission error to %s:%d: %s\n", inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port), strerror(errno));
+ #if 0
+ printf("Sent %d bytes of RTP data to %s:%d\n", res, inet_ntoa(rtp->them.sin_addr), ntohs(rtp->them.sin_port));
+ #endif
+ }
+ if (x ==0) {
+ /* Clear marker bit and increment seqno */
+ rtpheader[0] = htonl((2 << 30) | (101 << 16) | (rtp->seqno++));
+ /* Make duration 240 */
+ rtpheader[3] |= htonl((240));
+ /* Set the End bit for the last 3 */
+ rtpheader[3] |= htonl((1 << 23));
+ }
+ }
+ return 0;
+}
+
static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec)
{
unsigned int *rtpheader;
@@ -387,6 +509,12 @@ static int ast_rtp_raw_write(struct ast_rtp *rtp, struct ast_frame *f, int codec
case AST_FORMAT_G729A:
pred = rtp->lastts + f->datalen * 8;
break;
+ case AST_FORMAT_GSM:
+ pred = rtp->lastts + f->datalen * 20 / 33;
+ break;
+ case AST_FORMAT_G723_1:
+ pred = rtp->lastts + g723_samples(f->data, f->datalen);
+ break;
default:
ast_log(LOG_WARNING, "Not sure about timestamp format for codec format %d\n", f->subclass);
}
@@ -423,8 +551,12 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
int codec;
int hdrlen = 12;
- /* Make sure we have enough space for RTP header */
+
+ /* If we have no peer, return immediately */
+ if (!rtp->them.sin_addr.s_addr)
+ return 0;
+ /* Make sure we have enough space for RTP header */
if (_f->frametype != AST_FRAME_VOICE) {
ast_log(LOG_WARNING, "RTP can only send voice\n");
return -1;
@@ -436,6 +568,15 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
return -1;
}
+ if (rtp->lasttxformat != _f->subclass) {
+ /* New format, reset the smoother */
+ ast_log(LOG_DEBUG, "Ooh, format changed from %d to %d\n", rtp->lasttxformat, _f->subclass);
+ rtp->lasttxformat = _f->subclass;
+ if (rtp->smoother)
+ ast_smoother_free(rtp->smoother);
+ rtp->smoother = NULL;
+ }
+
switch(_f->subclass) {
case AST_FORMAT_ULAW:
@@ -465,7 +606,18 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
while((f = ast_smoother_read(rtp->smoother)))
ast_rtp_raw_write(rtp, f, codec);
break;
-
+ case AST_FORMAT_GSM:
+ if (!rtp->smoother) {
+ rtp->smoother = ast_smoother_new(33);
+ }
+ if (!rtp->smoother) {
+ ast_log(LOG_WARNING, "Unable to create GSM smoother :(\n");
+ return -1;
+ }
+ ast_smoother_feed(rtp->smoother, _f);
+ while((f = ast_smoother_read(rtp->smoother)))
+ ast_rtp_raw_write(rtp, f, codec);
+ break;
default:
ast_log(LOG_WARNING, "Not sure about sending format %d packets\n", _f->subclass);
if (_f->offset < hdrlen) {