Version 0.1.12 from FTP

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@461 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer 2002-06-16 16:06:38 +00:00
parent 738689ea86
commit ede7bca2a7
5 changed files with 471 additions and 6 deletions

14
CREDITS
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@ -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 ===

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@ -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

36
configs/sip.conf.sample Executable file
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@ -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

54
include/asterisk/rtp.h Executable file
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@ -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

364
rtp.c Executable file
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@ -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;
}