chan_sip.c

Asterisk中信道部分的源码 。。。。

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2006, Digium, Inc.
 *
 * Mark Spencer <[e-mail]>
 *
 * 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 Implementation of Session Initiation Protocol
 *
 * \author Mark Spencer <[e-mail]>
 *
 * See Also:
 * \arg \ref AstCREDITS
 *
 * Implementation of RFC 3261 - without S/MIME, TCP and TLS support
 * Configuration file \link Config_sip sip.conf \endlink
 *
 *
 * \todo SIP over TCP
 * \todo SIP over TLS
 * \todo Better support of forking
 * \todo VIA branch tag transaction checking
 * \todo Transaction support
 *
 * \ingroup channel_drivers
 *
 * \par Overview of the handling of SIP sessions
 * The SIP channel handles several types of SIP sessions, or dialogs,
 * not all of them being "telephone calls".
 * - Incoming calls that will be sent to the PBX core
 * - Outgoing calls, generated by the PBX
 * - SIP subscriptions and notifications of states and voicemail messages
 * - SIP registrations, both inbound and outbound
 * - SIP peer management (peerpoke, OPTIONS)
 * - SIP text messages
 *
 * In the SIP channel, there's a list of active SIP dialogs, which includes
 * all of these when they are active. "sip show channels" in the CLI will
 * show most of these, excluding subscriptions which are shown by
 * "sip show subscriptions"
 *
 * \par incoming packets
 * Incoming packets are received in the monitoring thread, then handled by
 * sipsock_read(). This function parses the packet and matches an existing
 * dialog or starts a new SIP dialog.
 * 
 * sipsock_read sends the packet to handle_request(), that parses a bit more.
 * if it's a response to an outbound request, it's sent to handle_response().
 * If it is a request, handle_request sends it to one of a list of functions
 * depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc
 * sipsock_read locks the ast_channel if it exists (an active call) and
 * unlocks it after we have processed the SIP message.
 *
 * A new INVITE is sent to handle_request_invite(), that will end up
 * starting a new channel in the PBX, the new channel after that executing
 * in a separate channel thread. This is an incoming "call".
 * When the call is answered, either by a bridged channel or the PBX itself
 * the sip_answer() function is called.
 *
 * The actual media - Video or Audio - is mostly handled by the RTP subsystem
 * in rtp.c 
 * 
 * \par Outbound calls
 * Outbound calls are set up by the PBX through the sip_request_call()
 * function. After that, they are activated by sip_call().
 * 
 * \par Hanging up
 * The PBX issues a hangup on both incoming and outgoing calls through
 * the sip_hangup() function
 *
 * \par Deprecated stuff
 * This is deprecated and will be removed after the 1.4 release
 * - the SIPUSERAGENT dialplan variable
 * - the ALERT_INFO dialplan variable
 */

/*** MODULEINFO
        <depend>res_features</depend>
 ***/


#include "asterisk.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision: 141809 $")

#include <stdio.h>
#include <ctype.h>
#include <string.h>
#include <unistd.h>
#include <sys/socket.h>
#include <sys/ioctl.h>
#include <net/if.h>
#include <errno.h>
#include <stdlib.h>
#include <fcntl.h>
#include <netdb.h>
#include <signal.h>
#include <sys/signal.h>
#include <netinet/in.h>
#include <netinet/in_systm.h>
#include <arpa/inet.h>
#include <netinet/ip.h>
#include <regex.h>

#include "asterisk/lock.h"
#include "asterisk/channel.h"
#include "asterisk/config.h"
#include "asterisk/logger.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/options.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp.h"
#include "asterisk/udptl.h"
#include "asterisk/acl.h"
#include "asterisk/manager.h"
#include "asterisk/callerid.h"
#include "asterisk/cli.h"
#include "asterisk/app.h"
#include "asterisk/musiconhold.h"
#include "asterisk/dsp.h"
#include "asterisk/features.h"
#include "asterisk/srv.h"
#include "asterisk/astdb.h"
#include "asterisk/causes.h"
#include "asterisk/utils.h"
#include "asterisk/file.h"
#include "asterisk/astobj.h"
#include "asterisk/devicestate.h"
#include "asterisk/linkedlists.h"
#include "asterisk/stringfields.h"
#include "asterisk/monitor.h"
#include "asterisk/localtime.h"
#include "asterisk/abstract_jb.h"
#include "asterisk/compiler.h"
#include "asterisk/threadstorage.h"
#include "asterisk/translate.h"

#ifndef FALSE
#define FALSE    0
#endif

#ifndef TRUE
#define TRUE     1
#endif

#define	SIPBUFSIZE		512

#define XMIT_ERROR		-2

#define VIDEO_CODEC_MASK        0x1fc0000       /*!< Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */
#ifndef IPTOS_MINCOST
#define IPTOS_MINCOST           0x02
#endif

/* #define VOCAL_DATA_HACK */

#define DEFAULT_DEFAULT_EXPIRY  120
#define DEFAULT_MIN_EXPIRY      60
#define DEFAULT_MAX_EXPIRY      3600
#define DEFAULT_REGISTRATION_TIMEOUT 20
#define DEFAULT_MAX_FORWARDS    "70"

/* guard limit must be larger than guard secs */
/* guard min must be < 1000, and should be >= 250 */
#define EXPIRY_GUARD_SECS       15                /*!< How long before expiry do we reregister */
#define EXPIRY_GUARD_LIMIT      30                /*!< Below here, we use EXPIRY_GUARD_PCT instead of 
                                                   EXPIRY_GUARD_SECS */
#define EXPIRY_GUARD_MIN        500                /*!< This is the minimum guard time applied. If 
                                                   GUARD_PCT turns out to be lower than this, it 
                                                   will use this time instead.
                                                   This is in milliseconds. */
#define EXPIRY_GUARD_PCT        0.20                /*!< Percentage of expires timeout to use when 
                                                    below EXPIRY_GUARD_LIMIT */
#define DEFAULT_EXPIRY 900                          /*!< Expire slowly */

static int min_expiry = DEFAULT_MIN_EXPIRY;        /*!< Minimum accepted registration time */
static int max_expiry = DEFAULT_MAX_EXPIRY;        /*!< Maximum accepted registration time */
static int default_expiry = DEFAULT_DEFAULT_EXPIRY;
static int expiry = DEFAULT_EXPIRY;

#ifndef MAX
#define MAX(a,b) ((a) > (b) ? (a) : (b))
#endif

#define CALLERID_UNKNOWN        "Unknown"

#define DEFAULT_MAXMS                2000             /*!< Qualification: Must be faster than 2 seconds by default */
#define DEFAULT_FREQ_OK              60 * 1000        /*!< Qualification: How often to check for the host to be up */
#define DEFAULT_FREQ_NOTOK           10 * 1000        /*!< Qualification: How often to check, if the host is down... */

#define DEFAULT_RETRANS              1000             /*!< How frequently to retransmit Default: 2 * 500 ms in RFC 3261 */
#define MAX_RETRANS                  6                /*!< Try only 6 times for retransmissions, a total of 7 transmissions */
#define SIP_TRANS_TIMEOUT            32000            /*!< SIP request timeout (rfc 3261) 64*T1 
                                                      \todo Use known T1 for timeout (peerpoke)
                                                      */
#define DEFAULT_TRANS_TIMEOUT        -1               /* Use default SIP transaction timeout */
#define MAX_AUTHTRIES                3                /*!< Try authentication three times, then fail */

#define SIP_MAX_HEADERS              64               /*!< Max amount of SIP headers to read */
#define SIP_MAX_LINES                64               /*!< Max amount of lines in SIP attachment (like SDP) */
#define SIP_MAX_PACKET               4096             /*!< Also from RFC 3261 (2543), should sub headers tho */

#define SDP_MAX_RTPMAP_CODECS        32               /*!< Maximum number of codecs allowed in received SDP */

#define INITIAL_CSEQ                 101              /*!< our initial sip sequence number */

/*! \brief 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;

static const char config[] = "sip.conf";
static const char notify_config[] = "sip_notify.conf";

#define RTP 	1
#define NO_RTP	0

/*! \brief Authorization scheme for call transfers 
\note Not a bitfield flag, since there are plans for other modes,
	like "only allow transfers for authenticated devices" */
enum transfermodes {
	TRANSFER_OPENFORALL,            /*!< Allow all SIP transfers */
	TRANSFER_CLOSED,                /*!< Allow no SIP transfers */
};


enum sip_result {
	AST_SUCCESS = 0,
	AST_FAILURE = -1,
};

/*! \brief States for the INVITE transaction, not the dialog 
	\note this is for the INVITE that sets up the dialog
*/
enum invitestates {
	INV_NONE = 0,	        /*!< No state at all, maybe not an INVITE dialog */
	INV_CALLING = 1,	/*!< Invite sent, no answer */
	INV_PROCEEDING = 2,	/*!< We got/sent 1xx message */
	INV_EARLY_MEDIA = 3,    /*!< We got 18x message with to-tag back */
	INV_COMPLETED = 4,	/*!< Got final response with error. Wait for ACK, then CONFIRMED */
	INV_CONFIRMED = 5,	/*!< Confirmed response - we've got an ack (Incoming calls only) */
	INV_TERMINATED = 6,	/*!< Transaction done - either successful (AST_STATE_UP) or failed, but done 
			    	     The only way out of this is a BYE from one side */
	INV_CANCELLED = 7,	/*!< Transaction cancelled by client or server in non-terminated state */
};

/* Do _NOT_ make any changes to this enum, or the array following it;
   if you think you are doing the right thing, you are probably
   not doing the right thing. If you think there are changes