chan_h323.c

Asterisk中信道部分的源码 。。。。

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2005
 *
 * OpenH323 Channel Driver for ASTERISK PBX.
 *			By Jeremy McNamara
 *                      For The NuFone Network
 *
 * chan_h323 has been derived from code created by
 *               Michael Manousos and Mark Spencer
 *
 * See http://www.asterisk.org for more information about
 * the Asterisk project. Please do not directly contact
 * any of the maintainers of this project for assistance;
 * the project provides a web site, mailing lists and IRC
 * channels for your use.
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License Version 2. See the LICENSE file
 * at the top of the source tree.
 */

/*! \file
 *
 * \brief This file is part of the chan_h323 driver for Asterisk
 *
 * \author Jeremy McNamara
 *
 * \par See also
 * \arg Config_h323
 *
 * \ingroup channel_drivers
 */

/*** MODULEINFO
	<depend>openh323</depend>
	<defaultenabled>yes</defaultenabled>
 ***/

#ifdef __cplusplus
extern "C" {
#endif

#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision: 123113 $")

#ifdef __cplusplus
}
#endif

#include <sys/types.h>
#include <sys/socket.h>
#include <sys/signal.h>
#include <sys/param.h>
#if defined(BSD) || defined(SOLARIS)
#ifndef IPTOS_MINCOST
#define IPTOS_MINCOST 0x02
#endif
#endif
#include <arpa/inet.h>
#include <net/if.h>
#include <netinet/in.h>
#include <netinet/in_systm.h>
#include <netinet/ip.h>
#include <unistd.h>
#include <stdlib.h>
#include <netdb.h>
#include <stdio.h>
#include <string.h>
#include <errno.h>
#include <fcntl.h>

#ifdef __cplusplus
extern "C" {
#endif

#include "asterisk/lock.h"
#include "asterisk/logger.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/module.h"
#include "asterisk/musiconhold.h"
#include "asterisk/pbx.h"
#include "asterisk/options.h"
#include "asterisk/utils.h"
#include "asterisk/lock.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp.h"
#include "asterisk/acl.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
#include "asterisk/dsp.h"
#include "asterisk/causes.h"
#include "asterisk/stringfields.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/astobj.h"

#ifdef __cplusplus
}
#endif

#include "h323/chan_h323.h"

receive_digit_cb on_receive_digit;
on_rtp_cb on_external_rtp_create;
start_rtp_cb on_start_rtp_channel;
setup_incoming_cb on_incoming_call;
setup_outbound_cb on_outgoing_call;
chan_ringing_cb	on_chan_ringing;
con_established_cb on_connection_established;
clear_con_cb on_connection_cleared;
answer_call_cb on_answer_call;
progress_cb on_progress;
rfc2833_cb on_set_rfc2833_payload;
hangup_cb on_hangup;
setcapabilities_cb on_setcapabilities;
setpeercapabilities_cb on_setpeercapabilities;

/* global debug flag */
int h323debug;

/*! Global jitterbuffer configuration - by default, jb is disabled */
static struct ast_jb_conf default_jbconf =
{
	.flags = 0,
	.max_size = -1,
	.resync_threshold = -1,
	.impl = ""
};
static struct ast_jb_conf global_jbconf;

/** Variables required by Asterisk */
static const char tdesc[] = "The NuFone Network's Open H.323 Channel Driver";
static const char config[] = "h323.conf";
static char default_context[AST_MAX_CONTEXT] = "default";
static struct sockaddr_in bindaddr;

#define GLOBAL_CAPABILITY (AST_FORMAT_G723_1 | AST_FORMAT_GSM | AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_G729A | AST_FORMAT_H261)

/** H.323 configuration values */
static int h323_signalling_port = 1720;
static char gatekeeper[100];
static int gatekeeper_disable = 1;
static int gatekeeper_discover = 0;
static int gkroute = 0;
/* Find user by alias (h.323 id) is default, alternative is the incomming call's source IP address*/
static int userbyalias = 1;
static int acceptAnonymous = 1;
static int tos = 0;
static char secret[50];
static unsigned int unique = 0;

static call_options_t global_options;

/** Private structure of a OpenH323 channel */
struct oh323_pvt {
	ast_mutex_t lock;					/* Channel private lock */
	call_options_t options;				/* Options to be used during call setup */
	int alreadygone;					/* Whether or not we've already been destroyed by our peer */
	int needdestroy;					/* if we need to be destroyed */
	call_details_t cd;					/* Call details */
	struct ast_channel *owner;			/* Who owns us */
	struct sockaddr_in sa;				/* Our peer */
	struct sockaddr_in redirip;			/* Where our RTP should be going if not to us */
	int nonCodecCapability;				/* non-audio capability */
	int outgoing;						/* Outgoing or incoming call? */
	char exten[AST_MAX_EXTENSION];		/* Requested extension */
	char context[AST_MAX_CONTEXT];		/* Context where to start */
	char accountcode[256];				/* Account code */
	char rdnis[80];						/* Referring DNIS, if available */
	int amaflags;						/* AMA Flags */
	struct ast_rtp *rtp;				/* RTP Session */
	struct ast_dsp *vad;				/* Used for in-band DTMF detection */
	int nativeformats;					/* Codec formats supported by a channel */
	int needhangup;						/* Send hangup when Asterisk is ready */
	int hangupcause;					/* Hangup cause from OpenH323 layer */
	int newstate;						/* Pending state change */
	int newcontrol;						/* Pending control to send */
	int newdigit;						/* Pending DTMF digit to send */
	int newduration;					/* Pending DTMF digit duration to send */
	int pref_codec;						/* Preferred codec */
	int peercapability;					/* Capabilities learned from peer */
	int jointcapability;				/* Common capabilities for local and remote side */
	struct ast_codec_pref peer_prefs;	/* Preferenced list of codecs which remote side supports */
	int dtmf_pt;						/* Payload code used for RFC2833 messages */
	int curDTMF;						/* DTMF tone being generated to Asterisk side */
	int DTMFsched;						/* Scheduler descriptor for DTMF */
	int update_rtp_info;				/* Configuration of fd's array is pending */
	int recvonly;						/* Peer isn't wish to receive our voice stream */
	int txDtmfDigit;					/* DTMF digit being to send to H.323 side */
	int noInbandDtmf;					/* Inband DTMF processing by DSP isn't available */
	int connection_established;			/* Call got CONNECT message */
	int got_progress;					/* Call got PROGRESS message, pass inband audio */
	struct oh323_pvt *next;				/* Next channel in list */
} *iflist = NULL;

static struct ast_user_list {
	ASTOBJ_CONTAINER_COMPONENTS(struct oh323_user);
} userl;

static struct ast_peer_list {
	ASTOBJ_CONTAINER_COMPONENTS(struct oh323_peer);
} peerl;

static struct ast_alias_list {
	ASTOBJ_CONTAINER_COMPONENTS(struct oh323_alias);
} aliasl;

/** Asterisk RTP stuff */
static struct sched_context *sched;
static struct io_context *io;

/** Protect the interface list (oh323_pvt) */
AST_MUTEX_DEFINE_STATIC(iflock);

/* Protect the monitoring thread, so only one process can kill or start it, and not
   when it's doing something critical. */
AST_MUTEX_DEFINE_STATIC(monlock);

/* Protect the H.323 capabilities list, to avoid more than one channel to set the capabilities simultaneaously in the h323 stack. */
AST_MUTEX_DEFINE_STATIC(caplock);

/* Protect the reload process */
AST_MUTEX_DEFINE_STATIC(h323_reload_lock);
static int h323_reloading = 0;

/* This is the thread for the monitor which checks for input on the channels
   which are not currently in use. */
static pthread_t monitor_thread = AST_PTHREADT_NULL;
static int restart_monitor(void);
static int h323_do_reload(void);

static struct ast_channel *oh323_request(const char *type, int format, void *data, int *cause);
static int oh323_digit_begin(struct ast_channel *c, char digit);
static int oh323_digit_end(struct ast_channel *c, char digit, unsigned int duration);
static int oh323_call(struct ast_channel *c, char *dest, int timeout);
static int oh323_hangup(struct ast_channel *c);
static int oh323_answer(struct ast_channel *c);
static struct ast_frame *oh323_read(struct ast_channel *c);
static int oh323_write(struct ast_channel *c, struct ast_frame *frame);
static int oh323_indicate(struct ast_channel *c, int condition, const void *data, size_t datalen);
static int oh323_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);

static const struct ast_channel_tech oh323_tech = {
	.type = "H323",
	.description = tdesc,
	.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
	.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER,
	.requester = oh323_request,
	.send_digit_begin = oh323_digit_begin,
	.send_digit_end = oh323_digit_end,
	.call = oh323_call,
	.hangup = oh323_hangup,
	.answer = oh323_answer,
	.read = oh323_read,
	.write = oh323_write,
	.indicate = oh323_indicate,
	.fixup = oh323_fixup,
	/* disable, for now */
#if 0
	.bridge = ast_rtp_bridge,
#endif
};

static const char* redirectingreason2str(int redirectingreason)
{
	switch (redirectingreason) {
	case 0:
		return "UNKNOWN";
	case 1:
		return "BUSY";
	case 2:
		return "NO_REPLY";
	case 0xF:
		return "UNCONDITIONAL";
	default:
		return "NOREDIRECT";
	}
}

static void oh323_destroy_alias(struct oh323_alias *alias)
{
	if (h323debug)
		ast_log(LOG_DEBUG, "Destroying alias '%s'\n", alias->name);
	free(alias);
}

static void oh323_destroy_user(struct oh323_user *user)
{
	if (h323debug)
		ast_log(LOG_DEBUG, "Destroying user '%s'\n", user->name);
	ast_free_ha(user->ha);
	free(user);
}

static void oh323_destroy_peer(struct oh323_peer *peer)
{
	if (h323debug)
		ast_log(LOG_DEBUG, "Destroying peer '%s'\n", peer->name);
	ast_free_ha(peer->ha);
	free(peer);
}

static int oh323_simulate_dtmf_end(const void *data)
{
	struct oh323_pvt *pvt = (struct oh323_pvt *)data;

	if (pvt) {
		ast_mutex_lock(&pvt->lock);
		/* Don't hold pvt lock while trying to lock the channel */
		while(pvt->owner && ast_channel_trylock(pvt->owner)) {
			ast_mutex_unlock(&pvt->lock);
			usleep(1);
			ast_mutex_lock(&pvt->lock);
		}

		if (pvt->owner) {
			struct ast_frame f = {
				.frametype = AST_FRAME_DTMF_END,
				.subclass = pvt->curDTMF,
				.samples = 0,
				.src = "SIMULATE_DTMF_END",
			};
			ast_queue_frame(pvt->owner, &f);
			ast_channel_unlock(pvt->owner);
		}

		pvt->DTMFsched = -1;
		ast_mutex_unlock(&pvt->lock);
	}

	return 0;
}

/* Channel and private structures should be already locked */
static void __oh323_update_info(struct ast_channel *c, struct oh323_pvt *pvt)
{
	if (c->nativeformats != pvt->nativeformats) {
		if (h323debug)
			ast_log(LOG_DEBUG, "Preparing %s for new native format\n", c->name);
		c->nativeformats = pvt->nativeformats;
		ast_set_read_format(c, c->readformat);
		ast_set_write_format(c, c->writeformat);
	}
	if (pvt->needhangup) {
		if (h323debug)
			ast_log(LOG_DEBUG, "Process pending hangup for %s\n", c->name);
		c->_softhangup |= AST_SOFTHANGUP_DEV;
		c->hangupcause = pvt->hangupcause;
		ast_queue_hangup(c);
		pvt->needhangup = 0;
		pvt->newstate = pvt->newcontrol = pvt->newdigit = pvt->DTMFsched = -1;
	}
	if (pvt->newstate >= 0) {
		ast_setstate(c, pvt->newstate);
		pvt->newstate = -1;
	}
	if (pvt->newcontrol >= 0) {
		ast_queue_control(c, pvt->newcontrol);
		pvt->newcontrol = -1;
	}
	if (pvt->newdigit >= 0) {
		struct ast_frame f = {
			.frametype = AST_FRAME_DTMF_END,
			.subclass = pvt->newdigit,
			.samples = pvt->newduration * 8,
			.len = pvt->newduration,
			.src = "UPDATE_INFO",
		};
		if (pvt->newdigit == ' ') {		/* signalUpdate message */
			f.subclass = pvt->curDTMF;
			if (pvt->DTMFsched >= 0) {
				AST_SCHED_DEL(sched, pvt->DTMFsched);
			}
		} else {						/* Regular input or signal message */
			if (pvt->newduration) {		/* This is a signal, signalUpdate follows */
				f.frametype = AST_FRAME_DTMF_BEGIN;
				AST_SCHED_DEL(sched, pvt->DTMFsched);
				pvt->DTMFsched = ast_sched_add(sched, pvt->newduration, oh323_simulate_dtmf_end, pvt);
				if (h323debug)
					ast_log(LOG_DTMF, "Scheduled DTMF END simulation for %d ms, id=%d\n", pvt->newduration, pvt->DTMFsched);
			}
			pvt->curDTMF = pvt->newdigit;
		}
		ast_queue_frame(c, &f);
		pvt->newdigit = -1;
	}
	if (pvt->update_rtp_info > 0) {
		if (pvt->rtp) {
			ast_jb_configure(c, &global_jbconf);
			c->fds[0] = ast_rtp_fd(pvt->rtp);
			c->fds[1] = ast_rtcp_fd(pvt->rtp);
			ast_queue_frame(pvt->owner, &ast_null_frame);	/* Tell Asterisk to apply changes */
		}
		pvt->update_rtp_info = -1;
	}
}

/* Only channel structure should be locked */
static void oh323_update_info(struct ast_channel *c)
{
	struct oh323_pvt *pvt = c->tech_pvt;

	if (pvt) {
		ast_mutex_lock(&pvt->lock);
		__oh323_update_info(c, pvt);
		ast_mutex_unlock(&pvt->lock);
	}
}

static void cleanup_call_details(call_details_t *cd)
{
	if (cd->call_token) {
		free(cd->call_token);
		cd->call_token = NULL;
	}
	if (cd->call_source_aliases) {
		free(cd->call_source_aliases);
		cd->call_source_aliases = NULL;
	}
	if (cd->call_dest_alias) {
		free(cd->call_dest_alias);
		cd->call_dest_alias = NULL;
	}
	if (cd->call_source_name) {
		free(cd->call_source_name);
		cd->call_source_name = NULL;
	}
	if (cd->call_source_e164) {
		free(cd->call_source_e164);
		cd->call_source_e164 = NULL;
	}
	if (cd->call_dest_e164) {
		free(cd->call_dest_e164);
		cd->call_dest_e164 = NULL;
	}
	if (cd->sourceIp) {
		free(cd->sourceIp);
		cd->sourceIp = NULL;
	}
	if (cd->redirect_number) {
		free(cd->redirect_number);