chan_sip.c

/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2005, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
 * 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
 * 
 * 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
 */


#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.h"

ASTERISK_FILE_VERSION(__FILE__, "$Revision$")

#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/lock.h"
#include "asterisk/sched.h"
#include "asterisk/io.h"
#include "asterisk/rtp.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/acl.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/dnsmgr.h"
#include "asterisk/devicestate.h"
#include "asterisk/linkedlists.h"

#ifdef OSP_SUPPORT
#include "asterisk/astosp.h"
#endif

#ifndef DEFAULT_USERAGENT
#define DEFAULT_USERAGENT "Asterisk PBX"
#endif
 
#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 SIPDUMPER
#define DEFAULT_DEFAULT_EXPIRY  120
#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 */

static int max_expiry = DEFAULT_MAX_EXPIRY;
static int default_expiry = DEFAULT_DEFAULT_EXPIRY;

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

#define CALLERID_UNKNOWN	"Unknown"



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

#define DEFAULT_RETRANS		1000		/* How frequently to retransmit */
						/* 2 * 500 ms in RFC 3261 */
#define MAX_RETRANS		6		/* Try only 6 times for retransmissions, a total of 7 transmissions */
#define MAX_AUTHTRIES		3		/* Try authentication three times, then fail */


#define DEBUG_READ	0			/* Recieved data	*/
#define DEBUG_SEND	1			/* Transmit data	*/

static const char desc[] = "Session Initiation Protocol (SIP)";
static const char channeltype[] = "SIP";
static const char config[] = "sip.conf";
static const char notify_config[] = "sip_notify.conf";

#define RTP 	1
#define NO_RTP	0

/* 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
   needed, get someone else to review them first _before_
   submitting a patch. If these two lists do not match properly
   bad things will happen.
*/

enum subscriptiontype { 
	NONE = 0,
	TIMEOUT,
	XPIDF_XML,
	DIALOG_INFO_XML,
	CPIM_PIDF_XML,
	PIDF_XML
};

static const struct cfsubscription_types {
	enum subscriptiontype type;
	const char * const event;
	const char * const mediatype;
	const char * const text;
} subscription_types[] = {
	{ NONE,            "-",        "unknown",	                  "unknown" },
 	/* IETF draft: draft-ietf-sipping-dialog-package-05.txt */
	{ DIALOG_INFO_XML, "dialog",   "application/dialog-info+xml", "dialog-info+xml" },
	{ CPIM_PIDF_XML,   "presence", "application/cpim-pidf+xml",   "cpim-pidf+xml" },  /* RFC 3863 */
	{ PIDF_XML,        "presence", "application/pidf+xml",        "pidf+xml" },       /* RFC 3863 */
	{ XPIDF_XML,       "presence", "application/xpidf+xml",       "xpidf+xml" }       /* Pre-RFC 3863 with MS additions */
};

enum sipmethod {
	SIP_UNKNOWN,
	SIP_RESPONSE,
	SIP_REGISTER,
	SIP_OPTIONS,
	SIP_NOTIFY,
	SIP_INVITE,
	SIP_ACK,
	SIP_PRACK,
	SIP_BYE,
	SIP_REFER,
	SIP_SUBSCRIBE,
	SIP_MESSAGE,
	SIP_UPDATE,
	SIP_INFO,
	SIP_CANCEL,
	SIP_PUBLISH,
} sip_method_list;

enum sip_auth_type {
	PROXY_AUTH,
	WWW_AUTH,
};

static const struct  cfsip_methods { 
	enum sipmethod id;
	int need_rtp;		/*!< when this is the 'primary' use for a pvt structure, does it need RTP? */
	char * const text;
} sip_methods[] = {
	{ SIP_UNKNOWN,	 RTP,    "-UNKNOWN-" },
	{ SIP_RESPONSE,	 NO_RTP, "SIP/2.0" },
	{ SIP_REGISTER,	 NO_RTP, "REGISTER" },
 	{ SIP_OPTIONS,	 NO_RTP, "OPTIONS" },
	{ SIP_NOTIFY,	 NO_RTP, "NOTIFY" },
	{ SIP_INVITE,	 RTP,    "INVITE" },
	{ SIP_ACK,	 NO_RTP, "ACK" },
	{ SIP_PRACK,	 NO_RTP, "PRACK" },
	{ SIP_BYE,	 NO_RTP, "BYE" },
	{ SIP_REFER,	 NO_RTP, "REFER" },
	{ SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE" },
	{ SIP_MESSAGE,	 NO_RTP, "MESSAGE" },
	{ SIP_UPDATE,	 NO_RTP, "UPDATE" },
	{ SIP_INFO,	 NO_RTP, "INFO" },
	{ SIP_CANCEL,	 NO_RTP, "CANCEL" },
	{ SIP_PUBLISH,	 NO_RTP, "PUBLISH" }
};

/*! \brief Structure for conversion between compressed SIP and "normal" SIP */
static const struct cfalias {
	char * const fullname;
	char * const shortname;
} aliases[] = {
	{ "Content-Type", "c" },
	{ "Content-Encoding", "e" },
	{ "From", "f" },
	{ "Call-ID", "i" },
	{ "Contact", "m" },
	{ "Content-Length", "l" },
	{ "Subject", "s" },
	{ "To", "t" },
	{ "Supported", "k" },
	{ "Refer-To", "r" },
	{ "Referred-By", "b" },
	{ "Allow-Events", "u" },
	{ "Event", "o" },
	{ "Via", "v" },
	{ "Accept-Contact",      "a" },
	{ "Reject-Contact",      "j" },
	{ "Request-Disposition", "d" },
	{ "Session-Expires",     "x" },
};

/*!  Define SIP option tags, used in Require: and Supported: headers 
 	We need to be aware of these properties in the phones to use 
	the replace: header. We should not do that without knowing
	that the other end supports it... 
	This is nothing we can configure, we learn by the dialog
	Supported: header on the REGISTER (peer) or the INVITE
	(other devices)
	We are not using many of these today, but will in the future.
	This is documented in RFC 3261
*/
#define SUPPORTED		1
#define NOT_SUPPORTED		0

#define SIP_OPT_REPLACES	(1 << 0)
#define SIP_OPT_100REL		(1 << 1)
#define SIP_OPT_TIMER		(1 << 2)
#define SIP_OPT_EARLY_SESSION	(1 << 3)
#define SIP_OPT_JOIN		(1 << 4)
#define SIP_OPT_PATH		(1 << 5)
#define SIP_OPT_PREF		(1 << 6)
#define SIP_OPT_PRECONDITION	(1 << 7)
#define SIP_OPT_PRIVACY		(1 << 8)
#define SIP_OPT_SDP_ANAT	(1 << 9)
#define SIP_OPT_SEC_AGREE	(1 << 10)
#define SIP_OPT_EVENTLIST	(1 << 11)
#define SIP_OPT_GRUU		(1 << 12)
#define SIP_OPT_TARGET_DIALOG	(1 << 13)

/*! \brief List of well-known SIP options. If we get this in a require,
   we should check the list and answer accordingly. */
static const struct cfsip_options {
	int id;			/*!< Bitmap ID */
	int supported;		/*!< Supported by Asterisk ? */
	char * const text;	/*!< Text id, as in standard */
} sip_options[] = {
	/* Replaces: header for transfer */
	{ SIP_OPT_REPLACES,	SUPPORTED,	"replaces" },	
	/* RFC3262: PRACK 100% reliability */
	{ SIP_OPT_100REL,	NOT_SUPPORTED,	"100rel" },	
	/* SIP Session Timers */
	{ SIP_OPT_TIMER,	NOT_SUPPORTED,	"timer" },
	/* RFC3959: SIP Early session support */
	{ SIP_OPT_EARLY_SESSION, NOT_SUPPORTED,	"early-session" },
	/* SIP Join header support */
	{ SIP_OPT_JOIN,		NOT_SUPPORTED,	"join" },
	/* RFC3327: Path support */
	{ SIP_OPT_PATH,		NOT_SUPPORTED,	"path" },
	/* RFC3840: Callee preferences */
	{ SIP_OPT_PREF,		NOT_SUPPORTED,	"pref" },
	/* RFC3312: Precondition support */
	{ SIP_OPT_PRECONDITION,	NOT_SUPPORTED,	"precondition" },
	/* RFC3323: Privacy with proxies*/
	{ SIP_OPT_PRIVACY,	NOT_SUPPORTED,	"privacy" },
	/* RFC4092: Usage of the SDP ANAT Semantics in the SIP */
	{ SIP_OPT_SDP_ANAT,	NOT_SUPPORTED,	"sdp-anat" },
	/* RFC3329: Security agreement mechanism */
	{ SIP_OPT_SEC_AGREE,	NOT_SUPPORTED,	"sec_agree" },
	/* SIMPLE events:  draft-ietf-simple-event-list-07.txt */
	{ SIP_OPT_EVENTLIST,	NOT_SUPPORTED,	"eventlist" },
	/* GRUU: Globally Routable User Agent URI's */
	{ SIP_OPT_GRUU,		NOT_SUPPORTED,	"gruu" },
	/* Target-dialog: draft-ietf-sip-target-dialog-00.txt */
	{ SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED,	"target-dialog" },
};


/*! \brief SIP Methods we support */
#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"

/*! \brief SIP Extensions we support */
#define SUPPORTED_EXTENSIONS "replaces" 

#define DEFAULT_SIP_PORT	5060	/*!< From RFC 3261 (former 2543) */
#define SIP_MAX_PACKET		4096	/*!< Also from RFC 3261 (2543), should sub headers tho */

static char default_useragent[AST_MAX_EXTENSION] = DEFAULT_USERAGENT;

#define DEFAULT_CONTEXT "default"
static char default_context[AST_MAX_CONTEXT] = DEFAULT_CONTEXT;
static char default_subscribecontext[AST_MAX_CONTEXT];

#define DEFAULT_VMEXTEN "asterisk"
static char global_vmexten[AST_MAX_EXTENSION] = DEFAULT_VMEXTEN;

static char default_language[MAX_LANGUAGE] = "";

#define DEFAULT_CALLERID "asterisk"
static char default_callerid[AST_MAX_EXTENSION] = DEFAULT_CALLERID;

static char default_fromdomain[AST_MAX_EXTENSION] = "";

#define DEFAULT_NOTIFYMIME "application/simple-message-summary"
static char default_notifymime[AST_MAX_EXTENSION] = DEFAULT_NOTIFYMIME;

static int global_notifyringing = 1;	/*!< Send notifications on ringing */

static int default_qualify = 0;		/*!< Default Qualify= setting */

static struct ast_flags global_flags = {0};		/*!< global SIP_ flags */
static struct ast_flags global_flags_page2 = {0};	/*!< more global SIP_ flags */

static int srvlookup = 0;		/*!< SRV Lookup on or off. Default is off, RFC behavior is on */

static int pedanticsipchecking = 0;	/*!< Extra checking ?  Default off */

static int autocreatepeer = 0;		/*!< Auto creation of peers at registration? Default off. */

static int relaxdtmf = 0;

static int global_rtptimeout = 0;

static int global_rtpholdtimeout = 0;

static int global_rtpkeepalive = 0;

static int global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;	
static int global_regattempts_max = 0;

/* Object counters */
static int suserobjs = 0;
static int ruserobjs = 0;
static int speerobjs = 0;
static int rpeerobjs = 0;
static int apeerobjs = 0;
static int regobjs = 0;

static int global_allowguest = 1;    /*!< allow unauthenticated users/peers to connect? */

#define DEFAULT_MWITIME 10
static int global_mwitime = DEFAULT_MWITIME;	/*!< Time between MWI checks for peers */

static int usecnt =0;
AST_MUTEX_DEFINE_STATIC(usecnt_lock);

AST_MUTEX_DEFINE_STATIC(rand_lock);

/*! \brief Protect the interface list (of sip_pvt's) */
AST_MUTEX_DEFINE_STATIC(iflock);

/*! \brief 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(netlock);

AST_MUTEX_DEFINE_STATIC(monlock);

/*! \brief 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);

/*! \brief Codecs that we support by default: */
static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263;
static int noncodeccapability = AST_RTP_DTMF;

static struct in_addr __ourip;
static struct sockaddr_in outboundproxyip;
static int ourport;

#define SIP_DEBUG_CONFIG 1 << 0
#define SIP_DEBUG_CONSOLE 1 << 1
static int sipdebug = 0;
static struct sockaddr_in debugaddr;

static int tos = 0;

static int videosupport = 0;

static int compactheaders = 0;				/*!< send compact sip headers */

static int recordhistory = 0;				/*!< Record SIP history. Off by default */
static int dumphistory = 0;				/*!< Dump history to verbose before destroying SIP dialog */

static char global_musicclass[MAX_MUSICCLASS] = "";	/*!< Global music on hold class */
#define DEFAULT_REALM	"asterisk"
static char global_realm[MAXHOSTNAMELEN] = DEFAULT_REALM; 	/*!< Default realm */
static char regcontext[AST_MAX_CONTEXT] = "";		/*!< Context for auto-extensions */

#define DEFAULT_EXPIRY 900				/*!< Expire slowly */
static int expiry = DEFAULT_EXPIRY;

static struct sched_context *sched;
static struct io_context *io;

#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 DEC_CALL_LIMIT	0
#define INC_CALL_LIMIT	1

static struct ast_codec_pref prefs;


/*! \brief sip_request: The data grabbed from the UDP socket */
struct sip_request {
	char *rlPart1; 		/*!< SIP Method Name or "SIP/2.0" protocol version */
	char *rlPart2; 		/*!< The Request URI or Response Status */
	int len;		/*!< Length */
	int headers;		/*!< # of SIP Headers */
	int method;		/*!< Method of this request */
	char *header[SIP_MAX_HEADERS];
	int lines;		/*!< SDP Content */
	char *line[SIP_MAX_LINES];
	char data[SIP_MAX_PACKET];
	int debug;		/*!< Debug flag for this packet */
	unsigned int flags;	/*!< SIP_PKT Flags for this packet */
};

struct sip_pkt;

/*! \brief Parameters to the transmit_invite function */
struct sip_invite_param {
	char *distinctive_ring;	/*!< Distinctive ring header */
	char *osptoken;		/*!< OSP token for this call */
	int addsipheaders;	/*!< Add extra SIP headers */
	char *uri_options;	/*!< URI options to add to the URI */
	char *vxml_url;		/*!< VXML url for Cisco phones */
	char *auth;		/*!< Authentication */
	char *authheader;	/*!< Auth header */
	enum sip_auth_type auth_type;	/*!< Authentication type */
};

struct sip_route {
	struct sip_route *next;
	char hop[0];
};

enum domain_mode {
	SIP_DOMAIN_AUTO,	/*!< This domain is auto-configured */
	SIP_DOMAIN_CONFIG,	/*!< This domain is from configuration */
};

struct domain {
	char domain[MAXHOSTNAMELEN];		/*!< SIP domain we are responsible for */
	char context[AST_MAX_EXTENSION];	/*!< Incoming context for this domain */
	enum domain_mode mode;			/*!< How did we find this domain? */
	AST_LIST_ENTRY(domain) list;		/*!< List mechanics */
};

static AST_LIST_HEAD_STATIC(domain_list, domain);	/*!< The SIP domain list */

int allow_external_domains;		/*!< Accept calls to external SIP domains? */

/*! \brief sip_history: Structure for saving transactions within a SIP dialog */
struct sip_history {
	char event[80];
	struct sip_history *next;
};

/*! \brief sip_auth: Creadentials for authentication to other SIP services */
struct sip_auth {
	char realm[AST_MAX_EXTENSION];  /*!< Realm in which these credentials are valid */
	char username[256];             /*!< Username */
	char secret[256];               /*!< Secret */
	char md5secret[256];            /*!< MD5Secret */
	struct sip_auth *next;          /*!< Next auth structure in list */
};

#define SIP_ALREADYGONE		(1 << 0)	/*!< Whether or not we've already been destroyed by our peer */
#define SIP_NEEDDESTROY		(1 << 1)	/*!< if we need to be destroyed */
#define SIP_NOVIDEO		(1 << 2)	/*!< Didn't get video in invite, don't offer */
#define SIP_RINGING		(1 << 3)	/*!< Have sent 180 ringing */
#define SIP_PROGRESS_SENT	(1 << 4)	/*!< Have sent 183 message progress */
#define SIP_NEEDREINVITE	(1 << 5)	/*!< Do we need to send another reinvite? */
#define SIP_PENDINGBYE		(1 << 6)	/*!< Need to send bye after we ack? */
#define SIP_GOTREFER		(1 << 7)	/*!< Got a refer? */
#define SIP_PROMISCREDIR	(1 << 8)	/*!< Promiscuous redirection */
#define SIP_TRUSTRPID		(1 << 9)	/*!< Trust RPID headers? */
#define SIP_USEREQPHONE		(1 << 10)	/*!< Add user=phone to numeric URI. Default off */
#define SIP_REALTIME		(1 << 11)	/*!< Flag for realtime users */
#define SIP_USECLIENTCODE	(1 << 12)	/*!< Trust X-ClientCode info message */
#define SIP_OUTGOING		(1 << 13)	/*!< Is this an outgoing call? */
#define SIP_SELFDESTRUCT	(1 << 14)	
#define SIP_DYNAMIC		(1 << 15)	/*!< Is this a dynamic peer? */
/* --- Choices for DTMF support in SIP channel */
#define SIP_DTMF		(3 << 16)	/*!< three settings, uses two bits */
#define SIP_DTMF_RFC2833	(0 << 16)	/*!< RTP DTMF */
#define SIP_DTMF_INBAND		(1 << 16)	/*!< Inband audio, only for ULAW/ALAW */
#define SIP_DTMF_INFO		(2 << 16)	/*!< SIP Info messages */
#define SIP_DTMF_AUTO		(3 << 16)	/*!< AUTO switch between rfc2833 and in-band DTMF */
/* NAT settings */
#define SIP_NAT			(3 << 18)	/*!< four settings, uses two bits */
#define SIP_NAT_NEVER		(0 << 18)	/*!< No nat support */
#define SIP_NAT_RFC3581		(1 << 18)
#define SIP_NAT_ROUTE		(2 << 18)
#define SIP_NAT_ALWAYS		(3 << 18)
/* re-INVITE related settings */
#define SIP_REINVITE		(3 << 20)	/*!< two bits used */
#define SIP_CAN_REINVITE	(1 << 20)	/*!< allow peers to be reinvited to send media directly p2p */
#define SIP_REINVITE_UPDATE	(2 << 20)	/*!< use UPDATE (RFC3311) when reinviting this peer */
/* "insecure" settings */
#define SIP_INSECURE_PORT	(1 << 22)	/*!< don't require matching port for incoming requests */
#define SIP_INSECURE_INVITE	(1 << 23)	/*!< don't require authentication for incoming INVITEs */
/* Sending PROGRESS in-band settings */
#define SIP_PROG_INBAND		(3 << 24)	/*!< three settings, uses two bits */
#define SIP_PROG_INBAND_NEVER	(0 << 24)
#define SIP_PROG_INBAND_NO	(1 << 24)
#define SIP_PROG_INBAND_YES	(2 << 24)
/* Open Settlement Protocol authentication */
#define SIP_OSPAUTH		(3 << 26)	/*!< four settings, uses two bits */
#define SIP_OSPAUTH_NO		(0 << 26)
#define SIP_OSPAUTH_GATEWAY	(1 << 26)
#define SIP_OSPAUTH_PROXY	(2 << 26)
#define SIP_OSPAUTH_EXCLUSIVE	(3 << 26)
/* Call states */
#define SIP_CALL_ONHOLD		(1 << 28)	 
#define SIP_CALL_LIMIT		(1 << 29)
/* Remote Party-ID Support */
#define SIP_SENDRPID		(1 << 30)

#define SIP_FLAGS_TO_COPY \
	(SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
	 SIP_PROG_INBAND | SIP_OSPAUTH | SIP_USECLIENTCODE | SIP_NAT | \
	 SIP_INSECURE_PORT | SIP_INSECURE_INVITE)

/* a new page of flags for peer */
#define SIP_PAGE2_RTCACHEFRIENDS	(1 << 0)
#define SIP_PAGE2_RTUPDATE		(1 << 1)
#define SIP_PAGE2_RTAUTOCLEAR		(1 << 2)
#define SIP_PAGE2_IGNOREREGEXPIRE	(1 << 3)
#define SIP_PAGE2_RT_FROMCONTACT 	(1 << 4)

/* SIP packet flags */
#define SIP_PKT_DEBUG		(1 << 0)	/*!< Debug this packet */
#define SIP_PKT_WITH_TOTAG	(1 << 1)	/*!< This packet has a to-tag */

static int global_rtautoclear = 120;

/*! \brief sip_pvt: PVT structures are used for each SIP conversation, ie. a call  */
static struct sip_pvt {
	ast_mutex_t lock;			/*!< Channel private lock */
	int method;				/*!< SIP method of this packet */
	char callid[80];			/*!< Global CallID */
	char randdata[80];			/*!< Random data */
	struct ast_codec_pref prefs;		/*!< codec prefs */
	unsigned int ocseq;			/*!< Current outgoing seqno */
	unsigned int icseq;			/*!< Current incoming seqno */
	ast_group_t callgroup;			/*!< Call group */
	ast_group_t pickupgroup;		/*!< Pickup group */
	int lastinvite;				/*!< Last Cseq of invite */
	unsigned int flags;			/*!< SIP_ flags */	
	int timer_t1;				/*!< SIP timer T1, ms rtt */
	unsigned int sipoptions;		/*!< Supported SIP sipoptions on the other end */
	int capability;				/*!< Special capability (codec) */
	int jointcapability;			/*!< Supported capability at both ends (codecs ) */
	int peercapability;			/*!< Supported peer capability */
	int prefcodec;				/*!< Preferred codec (outbound only) */
	int noncodeccapability;
	int callingpres;			/*!< Calling presentation */
	int authtries;				/*!< Times we've tried to authenticate */
	int expiry;				/*!< How long we take to expire */
	int branch;				/*!< One random number */
	char tag[11];				/*!< Another random number */
	int sessionid;				/*!< SDP Session ID */
	int sessionversion;			/*!< SDP Session Version */
	struct sockaddr_in sa;			/*!< Our peer */
	struct sockaddr_in redirip;		/*!< Where our RTP should be going if not to us */
	struct sockaddr_in vredirip;		/*!< Where our Video RTP should be going if not to us */
	int redircodecs;			/*!< Redirect codecs */
	struct sockaddr_in recv;		/*!< Received as */
	struct in_addr ourip;			/*!< Our IP */
	struct ast_channel *owner;		/*!< Who owns us */
	char exten[AST_MAX_EXTENSION];		/*!< Extension where to start */
	char refer_to[AST_MAX_EXTENSION];	/*!< Place to store REFER-TO extension */
	char referred_by[AST_MAX_EXTENSION];	/*!< Place to store REFERRED-BY extension */
	char refer_contact[AST_MAX_EXTENSION];	/*!< Place to store Contact info from a REFER extension */
	struct sip_pvt *refer_call;		/*!< Call we are referring */
	struct sip_route *route;		/*!< Head of linked list of routing steps (fm Record-Route) */
	int route_persistant;			/*!< Is this the "real" route? */
	char from[256];				/*!< The From: header */
	char useragent[256];			/*!< User agent in SIP request */
	char context[AST_MAX_CONTEXT];		/*!< Context for this call */
	char subscribecontext[AST_MAX_CONTEXT];	/*!< Subscribecontext */
	char fromdomain[MAXHOSTNAMELEN];	/*!< Domain to show in the from field */
	char fromuser[AST_MAX_EXTENSION];	/*!< User to show in the user field */
	char fromname[AST_MAX_EXTENSION];	/*!< Name to show in the user field */
	char tohost[MAXHOSTNAMELEN];		/*!< Host we should put in the "to" field */
	char language[MAX_LANGUAGE];		/*!< Default language for this call */
	char musicclass[MAX_MUSICCLASS];	/*!< Music on Hold class */
	char rdnis[256];			/*!< Referring DNIS */
	char theirtag[256];			/*!< Their tag */
	char username[256];			/*!< [user] name */
	char peername[256];			/*!< [peer] name, not set if [user] */
	char authname[256];			/*!< Who we use for authentication */
	char uri[256];				/*!< Original requested URI */
	char okcontacturi[256];			/*!< URI from the 200 OK on INVITE */
	char peersecret[256];			/*!< Password */
	char peermd5secret[256];
	struct sip_auth *peerauth;		/*!< Realm authentication */
	char cid_num[256];			/*!< Caller*ID */
	char cid_name[256];			/*!< Caller*ID */
	char via[256];				/*!< Via: header */
	char fullcontact[128];			/*!< The Contact: that the UA registers with us */
	char accountcode[AST_MAX_ACCOUNT_CODE];	/*!< Account code */
	char our_contact[256];			/*!< Our contact header */
	char *rpid;				/*!< Our RPID header */
	char *rpid_from;			/*!< Our RPID From header */
	char realm[MAXHOSTNAMELEN];		/*!< Authorization realm */
	char nonce[256];			/*!< Authorization nonce */
	int noncecount;				/*!< Nonce-count */
	char opaque[256];			/*!< Opaque nonsense */
	char qop[80];				/*!< Quality of Protection, since SIP wasn't complicated enough yet. */
	char domain[MAXHOSTNAMELEN];		/*!< Authorization domain */
	char lastmsg[256];			/*!< Last Message sent/received */
	int amaflags;				/*!< AMA Flags */
	int pendinginvite;			/*!< Any pending invite */
#ifdef OSP_SUPPORT
	int osphandle;				/*!< OSP Handle for call */
	time_t ospstart;			/*!< OSP Start time */
	unsigned int osptimelimit;		/*!< OSP call duration limit */
#endif
	struct sip_request initreq;		/*!< Initial request */
	
	int maxtime;				/*!< Max time for first response */
	int initid;				/*!< Auto-congest ID if appropriate */
	int autokillid;				/*!< Auto-kill ID */
	time_t lastrtprx;			/*!< Last RTP received */
	time_t lastrtptx;			/*!< Last RTP sent */
	int rtptimeout;				/*!< RTP timeout time */
	int rtpholdtimeout;			/*!< RTP timeout when on hold */
	int rtpkeepalive;			/*!< Send RTP packets for keepalive */
	enum subscriptiontype subscribed;	/*!< Is this call a subscription?  */
	int stateid;
	int laststate;                          /*!< Last known extension state */
	int dialogver;
	
	struct ast_dsp *vad;			/*!< Voice Activation Detection dsp */
	
	struct sip_peer *peerpoke;		/*!< If this calls is to poke a peer, which one */
	struct sip_registry *registry;		/*!< If this is a REGISTER call, to which registry */
	struct ast_rtp *rtp;			/*!< RTP Session */
	struct ast_rtp *vrtp;			/*!< Video RTP session */
	struct sip_pkt *packets;		/*!< Packets scheduled for re-transmission */
	struct sip_history *history;		/*!< History of this SIP dialog */
	struct ast_variable *chanvars;		/*!< Channel variables to set for call */
	struct sip_pvt *next;			/*!< Next call in chain */
	struct sip_invite_param *options;	/*!< Options for INVITE */
} *iflist = NULL;

#define FLAG_RESPONSE (1 << 0)
#define FLAG_FATAL (1 << 1)

/*! \brief sip packet - read in sipsock_read, transmitted in send_request */
struct sip_pkt {
	struct sip_pkt *next;			/*!< Next packet */
	int retrans;				/*!< Retransmission number */
	int method;				/*!< SIP method for this packet */
	int seqno;				/*!< Sequence number */
	unsigned int flags;			/*!< non-zero if this is a response packet (e.g. 200 OK) */
	struct sip_pvt *owner;			/*!< Owner call */
	int retransid;				/*!< Retransmission ID */
	int timer_a;				/*!< SIP timer A, retransmission timer */
	int timer_t1;				/*!< SIP Timer T1, estimated RTT or 500 ms */
	int packetlen;				/*!< Length of packet */
	char data[0];
};	

/*! \brief Structure for SIP user data. User's place calls to us */
struct sip_user {
	/* Users who can access various contexts */
	ASTOBJ_COMPONENTS(struct sip_user);
	char secret[80];		/*!< Password */
	char md5secret[80];		/*!< Password in md5 */
	char context[AST_MAX_CONTEXT];	/*!< Default context for incoming calls */
	char subscribecontext[AST_MAX_CONTEXT];	/* Default context for subscriptions */
	char cid_num[80];		/*!< Caller ID num */
	char cid_name[80];		/*!< Caller ID name */
	char accountcode[AST_MAX_ACCOUNT_CODE];	/* Account code */
	char language[MAX_LANGUAGE];	/*!< Default language for this user */
	char musicclass[MAX_MUSICCLASS];/*!< Music on Hold class */
	char useragent[256];		/*!< User agent in SIP request */
	struct ast_codec_pref prefs;	/*!< codec prefs */
	ast_group_t callgroup;		/*!< Call group */
	ast_group_t pickupgroup;	/*!< Pickup Group */
	unsigned int flags;		/*!< SIP flags */	
	unsigned int sipoptions;	/*!< Supported SIP options */
	struct ast_flags flags_page2;	/*!< SIP_PAGE2 flags */
	int amaflags;			/*!< AMA flags for billing */
	int callingpres;		/*!< Calling id presentation */
	int capability;			/*!< Codec capability */
	int inUse;			/*!< Number of calls in use */
	int call_limit;			/*!< Limit of concurrent calls */
	struct ast_ha *ha;		/*!< ACL setting */
	struct ast_variable *chanvars;	/*!< Variables to set for channel created by user */
};

/* Structure for SIP peer data, we place calls to peers if registered  or fixed IP address (host) */
struct sip_peer {
	ASTOBJ_COMPONENTS(struct sip_peer);	/*!< name, refcount, objflags,  object pointers */
					/*!< peer->name is the unique name of this object */
	char secret[80];		/*!< Password */
	char md5secret[80];		/*!< Password in MD5 */
	struct sip_auth *auth;		/*!< Realm authentication list */
	char context[AST_MAX_CONTEXT];	/*!< Default context for incoming calls */
	char subscribecontext[AST_MAX_CONTEXT];	/*!< Default context for subscriptions */
	char username[80];		/*!< Temporary username until registration */ 
	char accountcode[AST_MAX_ACCOUNT_CODE];	/*!< Account code */
	int amaflags;			/*!< AMA Flags (for billing) */
	char tohost[MAXHOSTNAMELEN];	/*!< If not dynamic, IP address */
	char regexten[AST_MAX_EXTENSION]; /*!< Extension to register (if regcontext is used) */
	char fromuser[80];		/*!< From: user when calling this peer */
	char fromdomain[MAXHOSTNAMELEN];	/*!< From: domain when calling this peer */
	char fullcontact[256];		/*!< Contact registered with us (not in sip.conf) */
	char cid_num[80];		/*!< Caller ID num */
	char cid_name[80];		/*!< Caller ID name */
	int callingpres;		/*!< Calling id presentation */
	int inUse;			/*!< Number of calls in use */
	int call_limit;			/*!< Limit of concurrent calls */
	char vmexten[AST_MAX_EXTENSION]; /*!< Dialplan extension for MWI notify message*/
	char mailbox[AST_MAX_EXTENSION]; /*!< Mailbox setting for MWI checks */
	char language[MAX_LANGUAGE];	/*!<  Default language for prompts */
	char musicclass[MAX_MUSICCLASS];/*!<  Music on Hold class */
	char useragent[256];		/*!<  User agent in SIP request (saved from registration) */
	struct ast_codec_pref prefs;	/*!<  codec prefs */
	int lastmsgssent;
	time_t	lastmsgcheck;		/*!<  Last time we checked for MWI */
	unsigned int flags;		/*!<  SIP flags */	
	unsigned int sipoptions;	/*!<  Supported SIP options */
	struct ast_flags flags_page2;	/*!<  SIP_PAGE2 flags */
	int expire;			/*!<  When to expire this peer registration */
	int capability;			/*!<  Codec capability */
	int rtptimeout;			/*!<  RTP timeout */
	int rtpholdtimeout;		/*!<  RTP Hold Timeout */
	int rtpkeepalive;		/*!<  Send RTP packets for keepalive */
	ast_group_t callgroup;		/*!<  Call group */
	ast_group_t pickupgroup;	/*!<  Pickup group */
	struct ast_dnsmgr_entry *dnsmgr;/*!<  DNS refresh manager for peer */
	struct sockaddr_in addr;	/*!<  IP address of peer */

	/* Qualification */
	struct sip_pvt *call;		/*!<  Call pointer */
	int pokeexpire;			/*!<  When to expire poke (qualify= checking) */
	int lastms;			/*!<  How long last response took (in ms), or -1 for no response */
	int maxms;			/*!<  Max ms we will accept for the host to be up, 0 to not monitor */
	struct timeval ps;		/*!<  Ping send time */
	
	struct sockaddr_in defaddr;	/*!<  Default IP address, used until registration */
	struct ast_ha *ha;		/*!<  Access control list */
	struct ast_variable *chanvars;	/*!<  Variables to set for channel created by user */
	int lastmsg;
};

AST_MUTEX_DEFINE_STATIC(sip_reload_lock);
static int sip_reloading = 0;

/* States for outbound registrations (with register= lines in sip.conf */
#define REG_STATE_UNREGISTERED		0
#define REG_STATE_REGSENT		1
#define REG_STATE_AUTHSENT		2
#define REG_STATE_REGISTERED   		3
#define REG_STATE_REJECTED	   	4
#define REG_STATE_TIMEOUT	   	5
#define REG_STATE_NOAUTH	   	6
#define REG_STATE_FAILED		7


/*! \brief sip_registry: Registrations with other SIP proxies */
struct sip_registry {
	ASTOBJ_COMPONENTS_FULL(struct sip_registry,1,1);
	int portno;			/*!<  Optional port override */
	char username[80];		/*!<  Who we are registering as */
	char authuser[80];		/*!< Who we *authenticate* as */
	char hostname[MAXHOSTNAMELEN];	/*!< Domain or host we register to */
	char secret[80];		/*!< Password in clear text */	
	char md5secret[80];		/*!< Password in md5 */
	char contact[256];		/*!< Contact extension */
	char random[80];
	int expire;			/*!< Sched ID of expiration */
	int regattempts;		/*!< Number of attempts (since the last success) */
	int timeout; 			/*!< sched id of sip_reg_timeout */
	int refresh;			/*!< How often to refresh */
	struct sip_pvt *call;		/*!< create a sip_pvt structure for each outbound "registration call" in progress */
	int regstate;			/*!< Registration state (see above) */
	int callid_valid;		/*!< 0 means we haven't chosen callid for this registry yet. */
	char callid[80];		/*!< Global CallID for this registry */
	unsigned int ocseq;		/*!< Sequence number we got to for REGISTERs for this registry */
	struct sockaddr_in us;		/*!< Who the server thinks we are */
 	
 					/* Saved headers */
 	char realm[MAXHOSTNAMELEN];	/*!< Authorization realm */
 	char nonce[256];		/*!< Authorization nonce */
 	char domain[MAXHOSTNAMELEN];	/*!< Authorization domain */
 	char opaque[256];		/*!< Opaque nonsense */
 	char qop[80];			/*!< Quality of Protection. */
	int noncecount;			/*!< Nonce-count */
 
 	char lastmsg[256];		/*!< Last Message sent/received */
};

/*! \brief  The user list: Users and friends ---*/
static struct ast_user_list {
	ASTOBJ_CONTAINER_COMPONENTS(struct sip_user);
} userl;

/*! \brief  The peer list: Peers and Friends ---*/
static struct ast_peer_list {
	ASTOBJ_CONTAINER_COMPONENTS(struct sip_peer);
} peerl;

/*! \brief  The register list: Other SIP proxys we register with and call ---*/
static struct ast_register_list {
	ASTOBJ_CONTAINER_COMPONENTS(struct sip_registry);
	int recheck;
} regl;


static int __sip_do_register(struct sip_registry *r);

static int sipsock  = -1;


static struct sockaddr_in bindaddr = { 0, };
static struct sockaddr_in externip;
static char externhost[MAXHOSTNAMELEN] = "";
static time_t externexpire = 0;
static int externrefresh = 10;
static struct ast_ha *localaddr;

/* The list of manual NOTIFY types we know how to send */
struct ast_config *notify_types;

static struct sip_auth *authl;          /*!< Authentication list */


static struct ast_frame  *sip_read(struct ast_channel *ast);
static int transmit_response(struct sip_pvt *p, char *msg, struct sip_request *req);
static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
static int transmit_response_with_unsupported(struct sip_pvt *p, char *msg, struct sip_request *req, char *unsupported);
static int transmit_response_with_auth(struct sip_pvt *p, char *msg, struct sip_request *req, char *rand, int reliable, char *header, int stale);
static int transmit_request(struct sip_pvt *p, int sipmethod, int inc, int reliable, int newbranch);
static int transmit_request_with_auth(struct sip_pvt *p, int sipmethod, int inc, int reliable, int newbranch);
static int transmit_invite(struct sip_pvt *p, int sipmethod, int sendsdp, int init);
static int transmit_reinvite_with_sdp(struct sip_pvt *p);
static int transmit_info_with_digit(struct sip_pvt *p, char digit);
static int transmit_info_with_vidupdate(struct sip_pvt *p);
static int transmit_message_with_text(struct sip_pvt *p, const char *text);
static int transmit_refer(struct sip_pvt *p, const char *dest);
static int sip_sipredirect(struct sip_pvt *p, const char *dest);
static struct sip_peer *temp_peer(const char *name);
static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req, char *header, char *respheader, int sipmethod, int init);
static void free_old_route(struct sip_route *route);
static int build_reply_digest(struct sip_pvt *p, int method, char *digest, int digest_len);
static int update_call_counter(struct sip_pvt *fup, int event);
static struct sip_peer *build_peer(const char *name, struct ast_variable *v, int realtime);
static struct sip_user *build_user(const char *name, struct ast_variable *v, int realtime);
static int sip_do_reload(void);
static int expire_register(void *data);
static int callevents = 0;

static struct ast_channel *sip_request_call(const char *type, int format, void *data, int *cause);
static int sip_devicestate(void *data);
static int sip_sendtext(struct ast_channel *ast, const char *text);
static int sip_call(struct ast_channel *ast, char *dest, int timeout);
static int sip_hangup(struct ast_channel *ast);
static int sip_answer(struct ast_channel *ast);
static struct ast_frame *sip_read(struct ast_channel *ast);
static int sip_write(struct ast_channel *ast, struct ast_frame *frame);
static int sip_indicate(struct ast_channel *ast, int condition);
static int sip_transfer(struct ast_channel *ast, const char *dest);
static int sip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
static int sip_senddigit(struct ast_channel *ast, char digit);
static int clear_realm_authentication(struct sip_auth *authlist);                            /* Clear realm authentication list (at reload) */
static struct sip_auth *add_realm_authentication(struct sip_auth *authlist, char *configuration, int lineno);   /* Add realm authentication in list */
static struct sip_auth *find_realm_authentication(struct sip_auth *authlist, char *realm);         /* Find authentication for a specific realm */
static int check_sip_domain(const char *domain, char *context, size_t len); /* Check if domain is one of our local domains */
static void append_date(struct sip_request *req);	/* Append date to SIP packet */
static int determine_firstline_parts(struct sip_request *req);
static void sip_dump_history(struct sip_pvt *dialog);	/* Dump history to LOG_DEBUG at end of dialog, before destroying data */
static const struct cfsubscription_types *find_subscription_type(enum subscriptiontype subtype);
static int transmit_state_notify(struct sip_pvt *p, int state, int full, int substate);
static char *gettag(struct sip_request *req, char *header, char *tagbuf, int tagbufsize);

/*! \brief Definition of this channel for PBX channel registration */
static const struct ast_channel_tech sip_tech = {
	.type = channeltype,
	.description = "Session Initiation Protocol (SIP)",
	.capabilities = ((AST_FORMAT_MAX_AUDIO << 1) - 1),
	.properties = AST_CHAN_TP_WANTSJITTER,
	.requester = sip_request_call,
	.devicestate = sip_devicestate,
	.call = sip_call,
	.hangup = sip_hangup,
	.answer = sip_answer,
	.read = sip_read,
	.write = sip_write,
	.write_video = sip_write,
	.indicate = sip_indicate,
	.transfer = sip_transfer,
	.fixup = sip_fixup,
	.send_digit = sip_senddigit,
	.bridge = ast_rtp_bridge,
	.send_text = sip_sendtext,
};

/*!
  \brief Thread-safe random number generator
  \return a random number

  This function uses a mutex lock to guarantee that no
  two threads will receive the same random number.
 */
static force_inline int thread_safe_rand(void)
{
	int val;

	ast_mutex_lock(&rand_lock);
	val = rand();
	ast_mutex_unlock(&rand_lock);
	
	return val;
}

/*! \brief  find_sip_method: Find SIP method from header
 * Strictly speaking, SIP methods are case SENSITIVE, but we don't check 
 * following Jon Postel's rule: Be gentle in what you accept, strict with what you send */
int find_sip_method(char *msg)
{
	int i, res = 0;
	
	if (ast_strlen_zero(msg))
		return 0;

	for (i = 1; (i < (sizeof(sip_methods) / sizeof(sip_methods[0]))) && !res; i++) {
		if (!strcasecmp(sip_methods[i].text, msg)) 
			res = sip_methods[i].id;
	}
	return res;
}

/*! \brief  parse_sip_options: Parse supported header in incoming packet */
unsigned int parse_sip_options(struct sip_pvt *pvt, char *supported)
{
	char *next = NULL;
	char *sep = NULL;
	char *temp = ast_strdupa(supported);
	int i;
	unsigned int profile = 0;

	if (ast_strlen_zero(supported) )
		return 0;

	if (option_debug > 2 && sipdebug)
		ast_log(LOG_DEBUG, "Begin: parsing SIP \"Supported: %s\"\n", supported);

	next = temp;
	while (next) {
		char res=0;
		if ( (sep = strchr(next, ',')) != NULL) {