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