From ede7bca2a7f0c64b0244626bbbbfbcc4dbd946eb Mon Sep 17 00:00:00 2001
From: Mark Spencer <markster@digium.com>
Date: Sun, 16 Jun 2002 16:06:38 +0000
Subject: [PATCH] Version 0.1.12 from FTP
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
CREDITS | 14 +-
channels/Makefile | 9 +-
configs/sip.conf.sample | 36 ++++
include/asterisk/rtp.h | 54 ++++++
rtp.c | 364 ++++++++++++++++++++++++++++++++++++++++
5 files changed, 471 insertions(+), 6 deletions(-)
create mode 100755 configs/sip.conf.sample
create mode 100755 include/asterisk/rtp.h
create mode 100755 rtp.c
diff --git a/CREDITS b/CREDITS
index fe8543042f..7f7ed249d8 100755
--- a/CREDITS
+++ b/CREDITS
@@ -1,7 +1,3 @@
-=== HARDWARE DONORS ===
-* Thanks to QuickNet Technologies for their donation of an Internet
-PhoneJack and Linejack card to the project. (http://www.quicknet.net)
-
=== DEVELOPMENT SUPPORT ===
I'd like to thank the following companies for helping fund development of
@@ -9,6 +5,16 @@ Asterisk:
Pilosoft, Inc. - for supporting ADSI development in Asterisk
+GFS - for supporting ALSA development
+
+Telesthetic - for supporting SIP development
+
+=== HARDWARE DONORS ===
+* Thanks to QuickNet Technologies for their donation of an Internet
+PhoneJack and Linejack card to the project. (http://www.quicknet.net)
+
+=== MISCELLANEOUS PATCHES ===
+Oliver Daudey - ISDN4Linux fixes
=== OTHER SOURCE CODE IN ASTERISK ===
diff --git a/channels/Makefile b/channels/Makefile
index ca91618eb4..dc5cabba42 100755
--- a/channels/Makefile
+++ b/channels/Makefile
@@ -11,10 +11,15 @@
# the GNU General Public License
#
-CHANNEL_LIBS=chan_vofr.so chan_modem.so \
- chan_modem_aopen.so chan_iax.so chan_oss.so \
+CHANNEL_LIBS=chan_modem.so chan_iax.so chan_sip.so \
+ chan_modem_aopen.so chan_oss.so \
chan_modem_bestdata.so chan_modem_i4l.so
+#
+# If you really want VoFR you can have it :-P
+#
+#CHANNEL_LIBS+=chan_vofr
+
CHANNEL_LIBS+=$(shell [ -f /usr/include/linux/ixjuser.h ] && echo chan_phone.so)
CFLAGS+=-Wno-missing-prototypes -Wno-missing-declarations
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
new file mode 100755
index 0000000000..992aa07b15
--- /dev/null
+++ b/configs/sip.conf.sample
@@ -0,0 +1,36 @@
+;
+; SIP Configuration for Asterisk
+;
+[general]
+port = 5060 ; Port to bind to
+bindaddr = 0.0.0.0 ; Address to bind to
+context = default ; Default for incoming calls
+
+;[snomsip]
+;type=friend
+;secret=blah
+;host=dynamic
+;defaultip=192.168.0.59
+
+;[pingtel]
+;type=friend
+;insecure=yes ; Pingtel sends from different portno
+;username=pingtel
+;secret=blah
+;host=dynamic
+;defaultip=192.168.0.60
+
+;[cisco]
+;type=friend
+;username=cisco
+;secret=blah
+;host=dynamic
+;defaultip=192.168.0.4
+
+;[cisco1]
+;type=friend
+;insecure=yes
+;username=cisco1
+;secret=blah
+;host=dynamic
+;defaultip=192.168.0.4
diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
new file mode 100755
index 0000000000..7c66a85dab
--- /dev/null
+++ b/include/asterisk/rtp.h
@@ -0,0 +1,54 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Real-time Transport Protocol support
+ *
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#ifndef _ASTERISK_RTP_H
+#define _ASTERISK_RTP_H
+
+#include <asterisk/frame.h>
+#include <asterisk/io.h>
+#include <asterisk/sched.h>
+
+#include <netinet/in.h>
+
+#if defined(__cplusplus) || defined(c_plusplus)
+extern "C" {
+#endif
+
+struct ast_rtp;
+
+typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
+
+struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io);
+
+void ast_rtp_set_peer(struct ast_rtp *rtp, struct sockaddr_in *them);
+
+void ast_rtp_get_us(struct ast_rtp *rtp, struct sockaddr_in *us);
+
+void ast_rtp_destroy(struct ast_rtp *rtp);
+
+void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback);
+
+void ast_rtp_set_data(struct ast_rtp *rtp, void *data);
+
+int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *f);
+
+int ast2rtp(int id);
+
+int rtp2ast(int id);
+
+
+#if defined(__cplusplus) || defined(c_plusplus)
+}
+#endif
+
+#endif
diff --git a/rtp.c b/rtp.c
new file mode 100755
index 0000000000..3a87ad43fd
--- /dev/null
+++ b/rtp.c
@@ -0,0 +1,364 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Real-time Protocol Support
+ *
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <pthread.h>
+#include <string.h>
+#include <sys/time.h>
+#include <signal.h>
+#include <errno.h>
+#include <unistd.h>
+#include <netinet/in.h>
+#include <sys/time.h>
+#include <sys/socket.h>
+#include <arpa/inet.h>
+#include <fcntl.h>
+
+#include <asterisk/rtp.h>
+#include <asterisk/logger.h>
+#include <asterisk/options.h>
+#include <asterisk/channel.h>
+
+static int dtmftimeout = 300; /* 300 samples */
+
+struct ast_rtp {
+ int s;
+ char resp;
+ struct ast_frame f;
+ unsigned char rawdata[1024 + AST_FRIENDLY_OFFSET];
+ unsigned int ssrc;
+ unsigned int lastts;
+ unsigned int lastrxts;
+ int dtmfcount;
+ struct sockaddr_in us;
+ struct sockaddr_in them;
+ struct timeval rxcore;
+ struct timeval txcore;
+ int *ioid;
+ unsigned short seqno;
+ struct sched_context *sched;
+ struct io_context *io;
+ void *data;
+ ast_rtp_callback callback;
+};
+
+
+void ast_rtp_set_data(struct ast_rtp *rtp, void *data)
+{
+ rtp->data = data;
+}
+
+void ast_rtp_set_callback(struct ast_rtp *rtp, ast_rtp_callback callback)
+{
+ rtp->callback = callback;
+}
+
+static void send_dtmf(struct ast_rtp *rtp)
+{
+ printf("Sending dtmf: %d (%c)\n", rtp->resp, rtp->resp);
+ rtp->f.frametype = AST_FRAME_DTMF;
+ rtp->f.subclass = rtp->resp;
+ rtp->f.datalen = 0;
+ rtp->f.timelen = 0;
+ rtp->f.mallocd = 0;
+ rtp->f.src = "RTP";
+ rtp->resp = 0;
+ if (rtp->callback)
+ rtp->callback(rtp, &rtp->f, rtp->data);
+
+}
+
+static void process_rfc2833(struct ast_rtp *rtp, unsigned char *data, int len)
+{
+ unsigned int event;
+ char resp = 0;
+ event = ntohl(*((unsigned int *)(data)));
+ event >>= 24;
+#if 0
+ printf("Event: %08x (len = %d)\n", event, len);
+#endif
+ if (event < 10) {
+ resp = '0' + event;
+ } else if (event < 11) {
+ resp = '*';
+ } else if (event < 12) {
+ resp = '#';
+ } else if (event < 16) {
+ resp = 'A' + (event - 12);
+ }
+ if (rtp->resp && (rtp->resp != resp)) {
+ send_dtmf(rtp);
+ }
+ rtp->resp = resp;
+ rtp->dtmfcount = dtmftimeout;
+}
+
+static int rtpread(int *id, int fd, short events, void *cbdata)
+{
+ struct ast_rtp *rtp = cbdata;
+ int res;
+ struct sockaddr_in sin;
+ int len;
+ unsigned int seqno;
+ int payloadtype;
+ int hdrlen = 12;
+ unsigned int timestamp;
+
+ unsigned int *rtpheader;
+
+ len = sizeof(sin);
+
+ res = recvfrom(rtp->s, rtp->rawdata + AST_FRIENDLY_OFFSET, sizeof(rtp->rawdata) - AST_FRIENDLY_OFFSET,
+ 0, (struct sockaddr *)&sin, &len);
+
+ rtpheader = (unsigned int *)(rtp->rawdata + AST_FRIENDLY_OFFSET);
+ if (res < 0) {
+ ast_log(LOG_WARNING, "RTP Read error: %s\n", strerror(errno));
+ if (errno == EBADF)
+ CRASH;
+ return 1;
+ }
+ if (res < hdrlen) {
+ ast_log(LOG_WARNING, "RTP Read too short\n");
+ return 1;
+ }
+ /* Get fields */
+ seqno = ntohl(rtpheader[0]);
+ payloadtype = (seqno & 0x7f0000) >> 16;
+ seqno &= 0xffff;
+ timestamp = ntohl(rtpheader[1]);
+#if 0
+ printf("Got RTP packet from %s:%d (type %d, seq %d, ts %d, len = %d)\n", inet_ntoa(sin.sin_addr), ntohs(sin.sin_port), payloadtype, seqno, timestamp,res - hdrlen);
+#endif
+ rtp->f.frametype = AST_FRAME_VOICE;
+ rtp->f.subclass = rtp2ast(payloadtype);
+ if (rtp->f.subclass < 0) {
+ if (payloadtype == 101) {
+ /* It's special -- rfc2833 process it */
+ process_rfc2833(rtp, rtp->rawdata + AST_FRIENDLY_OFFSET + hdrlen, res - hdrlen);
+ } else
+ ast_log(LOG_NOTICE, "Unknown RTP codec %d received\n", payloadtype);
+ return 1;
+ }
+
+ if (!rtp->lastrxts)
+ rtp->lastrxts = timestamp;
+
+ if (rtp->dtmfcount) {
+#if 0
+ printf("dtmfcount was %d\n", rtp->dtmfcount);
+#endif
+ rtp->dtmfcount -= (timestamp - rtp->lastrxts);
+ if (rtp->dtmfcount < 0)
+ rtp->dtmfcount = 0;
+#if 0
+ if (dtmftimeout != rtp->dtmfcount)
+ printf("dtmfcount is %d\n", rtp->dtmfcount);
+#endif
+ }
+ rtp->lastrxts = timestamp;
+
+ /* Send any pending DTMF */
+ if (rtp->resp && !rtp->dtmfcount) {
+ send_dtmf(rtp);
+ /* Setup the voice frame again */
+ rtp->f.frametype = AST_FRAME_VOICE;
+ rtp->f.subclass = rtp2ast(payloadtype);
+ }
+ rtp->f.mallocd = 0;
+ rtp->f.datalen = res - hdrlen;
+ rtp->f.data = rtp->rawdata + hdrlen + AST_FRIENDLY_OFFSET;
+ rtp->f.offset = hdrlen + AST_FRIENDLY_OFFSET;
+ switch(rtp->f.subclass) {
+ case AST_FORMAT_ULAW:
+ case AST_FORMAT_ALAW:
+ rtp->f.timelen = rtp->f.datalen / 8;
+ break;
+ case AST_FORMAT_SLINEAR:
+ rtp->f.timelen = rtp->f.datalen / 16;
+ break;
+ case AST_FORMAT_GSM:
+ rtp->f.timelen = 20 * (rtp->f.datalen / 33);
+ break;
+ case AST_FORMAT_ADPCM:
+ rtp->f.timelen = rtp->f.datalen / 8;
+ break;
+ default:
+ ast_log(LOG_NOTICE, "Unable to calculate timelen for format %d\n", rtp->f.subclass);
+ break;
+ }
+ rtp->f.src = "RTP";
+ if (rtp->callback)
+ rtp->callback(rtp, &rtp->f, rtp->data);
+ return 1;
+}
+
+static struct {
+ int rtp;
+ int ast;
+} cmap[] = {
+ { 0, AST_FORMAT_ULAW },
+ { 3, AST_FORMAT_GSM },
+ { 4, AST_FORMAT_G723_1 },
+ { 5, AST_FORMAT_ADPCM },
+ { 8, AST_FORMAT_ALAW },
+ { 18, AST_FORMAT_G729A },
+};
+
+int rtp2ast(int id)
+{
+ int x;
+ for (x=0;x<sizeof(cmap) / sizeof(cmap[0]); x++) {
+ if (cmap[x].rtp == id)
+ return cmap[x].ast;
+ }
+ return -1;
+}
+
+int ast2rtp(int id)
+{
+ int x;
+ for (x=0;x<sizeof(cmap) / sizeof(cmap[0]); x++) {
+ if (cmap[x].ast == id)
+ return cmap[x].rtp;
+ }
+ return -1;
+}
+
+struct ast_rtp *ast_rtp_new(struct sched_context *sched, struct io_context *io)
+{
+ struct ast_rtp *rtp;
+ int x;
+ int flags;
+ rtp = malloc(sizeof(struct ast_rtp));
+ if (!rtp)
+ return NULL;
+ memset(rtp, 0, sizeof(struct ast_rtp));
+ rtp->them.sin_family = AF_INET;
+ rtp->us.sin_family = AF_INET;
+ rtp->s = socket(AF_INET, SOCK_DGRAM, 0);
+ rtp->ssrc = rand();
+ rtp->seqno = rand() & 0xffff;
+ if (rtp->s < 0) {
+ free(rtp);
+ ast_log(LOG_WARNING, "Unable to allocate socket: %s\n", strerror(errno));
+ return NULL;
+ }
+ flags = fcntl(rtp->s, F_GETFL);
+ fcntl(rtp->s, F_SETFL, flags | O_NONBLOCK);
+ for (;;) {
+ /* Find us a place */
+ x = (rand() % (65000-1025)) + 1025;
+ /* Must be an even port number by RTP spec */
+ x = x & ~1;
+ rtp->us.sin_port = htons(x);
+ if (!bind(rtp->s, &rtp->us, sizeof(rtp->us)))
+ break;
+ if (errno != EADDRINUSE) {
+ ast_log(LOG_WARNING, "Unexpected bind error: %s\n", strerror(errno));
+ close(rtp->s);
+ free(rtp);
+ return NULL;
+ }
+ }
+ rtp->io = io;
+ rtp->sched = sched;
+ rtp->ioid = ast_io_add(rtp->io, rtp->s, rtpread, AST_IO_IN, rtp);
+ return rtp;
+}
+
+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_us(struct ast_rtp *rtp, struct sockaddr_in *us)
+{
+ memcpy(us, &rtp->us, sizeof(rtp->us));
+}
+
+void ast_rtp_destroy(struct ast_rtp *rtp)
+{
+ if (rtp->ioid)
+ ast_io_remove(rtp->io, rtp->ioid);
+ if (rtp->s > -1)
+ close(rtp->s);
+ free(rtp);
+}
+
+static unsigned int calc_txstamp(struct ast_rtp *rtp)
+{
+ struct timeval now;
+ unsigned int ms;
+ if (!rtp->txcore.tv_sec && !rtp->txcore.tv_usec) {
+ gettimeofday(&rtp->txcore, NULL);
+ }
+ gettimeofday(&now, NULL);
+ ms = (now.tv_sec - rtp->txcore.tv_sec) * 1000;
+ ms += (now.tv_usec - rtp->txcore.tv_usec);
+ return ms;
+}
+
+int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
+{
+ int hdrlen = 12;
+ struct ast_frame *f;
+ int codec;
+ int res;
+ unsigned int ms;
+ unsigned int *rtpheader;
+
+ /* 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;
+ }
+
+ codec = ast2rtp(_f->subclass);
+ if (codec < 0) {
+ ast_log(LOG_WARNING, "Don't know how to send format %d packets with RTP\n", _f->subclass);
+ return -1;
+ }
+
+ if (_f->offset < hdrlen) {
+ f = ast_frdup(_f);
+ } else
+ f = _f;
+
+ ms = calc_txstamp(rtp) * 8;
+ switch(f->subclass) {
+ case AST_FORMAT_ULAW:
+ case AST_FORMAT_ALAW:
+ break;
+ default:
+ ast_log(LOG_WARNING, "Not sure about timestamp format for codec format %d\n", f->subclass);
+ }
+ rtp->lastts += f->datalen;
+ /* Get a pointer to the header */
+ rtpheader = (unsigned int *)(f->data - hdrlen);
+ rtpheader[0] = htonl((2 << 30) | (codec << 16) | (rtp->seqno++));
+ rtpheader[1] = htonl(rtp->lastts);
+ rtpheader[2] = htonl(rtp->ssrc);
+ if (rtp->them.sin_port && rtp->them.sin_addr.s_addr) {
+ res = sendto(rtp->s, (void *)rtpheader, f->datalen + hdrlen, 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
+ }
+ return 0;
+}
--
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