chan_sip.c

/*
 * Asterisk -- A telephony toolkit for Linux.
 *
 * Implementation of Session Initiation Protocol
 * 
 * Copyright (C) 2004, Digium, Inc.
 *
 * Mark Spencer <markster@digium.com>
 *
 * This program is free software, distributed under the terms of
 * the GNU General Public License
 */


#include <stdio.h>
#include <ctype.h>
#include <string.h>
#include <asterisk/lock.h>
#include <asterisk/channel.h>
#include <asterisk/channel_pvt.h>
#include <asterisk/config_pvt.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/md5.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>
#ifdef OSP_SUPPORT
#include <asterisk/astosp.h>
#endif
#include <sys/socket.h>
#include <sys/ioctl.h>
#include <net/if.h>
#include <errno.h>
#include <unistd.h>
#include <stdlib.h>
#include <fcntl.h>
#include <netdb.h>
#include <arpa/inet.h>
#include <signal.h>
#include <sys/signal.h>
#include <netinet/in_systm.h>
#include <netinet/ip.h>

#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

/* 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"

/* --- Choices for DTMF support in SIP channel */
#define SIP_DTMF_RFC2833	(1 << 0)	/* RTP DTMF */
#define SIP_DTMF_INBAND		(1 << 1)	/* Inband audio, only for ULAW/ALAW */
#define SIP_DTMF_INFO		(1 << 2)	/* SIP Info messages */


#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 */
#define MAX_RETRANS		5		/* Try only 5 times for retransmissions */

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

static char *desc = "Session Initiation Protocol (SIP)";
static char *channeltype = "SIP";
static char *tdesc = "Session Initiation Protocol (SIP)";
static char *config = "sip.conf";

#define DEFAULT_SIP_PORT	5060	/* From RFC 2543 */
#define SIP_MAX_PACKET		4096	/* Also from RFC 2543, should sub headers tho */

#define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER"

static char default_useragent[AST_MAX_EXTENSION] = DEFAULT_USERAGENT;

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

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 struct ast_flags global_flags = {0};		/* 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_progressinband = 0;

static int global_reg_timeout = DEFAULT_REGISTRATION_TIMEOUT;

/* 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;

#ifdef OSP_SUPPORT
static int global_ospauth = 0;		/* OSP = Open Settlement Protocol */
#endif

#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);


/* Protect the interface list (of sip_pvt's) */
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(netlock);

AST_MUTEX_DEFINE_STATIC(monlock);

/* 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);

/* 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 char ourhost[256];
static struct in_addr __ourip;
static struct sockaddr_in outboundproxyip;
static int ourport;

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 global_dtmfmode = SIP_DTMF_RFC2833;		/* DTMF mode default */
static int recordhistory = 0;				/* Record SIP history. Off by default */

static char global_musicclass[MAX_LANGUAGE] = "";	/* Global music on hold class */
static char global_realm[AST_MAX_EXTENSION] = "asterisk"; 	/* Default realm */
static char regcontext[AST_MAX_EXTENSION] = "";		/* Context for auto-extensions */

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

static struct sched_context *sched;
static struct io_context *io;
/* The private structures of the  sip channels are linked for
   selecting outgoing channels */
   
#define SIP_MAX_HEADERS		64
#define SIP_MAX_LINES 		64

#define DEC_IN_USE	0
#define INC_IN_USE	1
#define DEC_OUT_USE	2
#define INC_OUT_USE	3

static struct ast_codec_pref prefs;


/* 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 */
	char *header[SIP_MAX_HEADERS];
	int lines;						/* SDP Content */
	char *line[SIP_MAX_LINES];
	char data[SIP_MAX_PACKET];
};

struct sip_pkt;

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

struct sip_history {
	char event[80];
	struct sip_history *next;
};

#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		(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? */

/* sip_pvt: PVT structures are used for each SIP conversation, ie. a call  */
static struct sip_pvt {
	ast_mutex_t lock;			/* Channel private lock */
	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 */
	unsigned int callgroup;			/* Call group */
	unsigned int pickupgroup;		/* Pickup group */
	int lastinvite;				/* Last Cseq of invite */
	int flags;				/* SIP_ flags */	
	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 insecure;				/* Don't check source port/ip */
	int expiry;				/* How long we take to expire */
	int branch;				/* One random number */
	int canreinvite;			/* Do we support reinvite */
	int tag;				/* Another random number */
	int nat;				/* Whether to try to support NAT */
	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_EXTENSION];	/* Context for this call */
	char fromdomain[AST_MAX_EXTENSION];	/* Domain to show in the from field */
	char fromuser[AST_MAX_EXTENSION];	/* Domain to show in the user field */
	char tohost[AST_MAX_EXTENSION];		/* Host we should put in the "to" field */
	char language[MAX_LANGUAGE];		/* Default language for this call */
	char musicclass[MAX_LANGUAGE];          /* Music on Hold class */
	char rdnis[256];			/* Referring DNIS */
	char theirtag[256];			/* Their tag */
	char username[256];
	char peername[256];
	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];
	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[20];			/* Account code */
	char our_contact[256];			/* Our contact header */
	char realm[256];			/* Authorization realm */
	char nonce[256];			/* Authorization nonce */
	char opaque[256];			/* Opaque nonsense */
	char qop[80];				/* Quality of Protection, since SIP wasn't complicated enough yet. */
	char domain[256];			/* Authorization domain */
	char lastmsg[256];			/* Last Message sent/received */
	int amaflags;				/* AMA Flags */
	int pendinginvite;			/* Any pending invite */
#ifdef OSP_SUPPORT
	int ospauth;				/* Allow OSP Authentication */
	int osphandle;				/* OSP Handle for call */
	time_t ospstart;			/* OSP Start time */
#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 */
	int rtptimeout;				/* RTP timeout time */
	int rtpholdtimeout;			/* RTP timeout when on hold */

	int subscribed;				/* Is this call a subscription?  */
    	int stateid;
	int dialogver;
	
	int progressinband;
	
	int dtmfmode;				/* DTMF to use for this call */
	struct ast_dsp *vad;
	
	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 *vars;
	struct sip_pvt *next;			/* Next call in chain */
} *iflist = NULL;

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

/* sip packet - read in sipsock_read, transmitted in send_request */
struct sip_pkt {
	struct sip_pkt *next;				/* Next packet */
	int retrans;					/* Retransmission number */
	int seqno;					/* Sequence number */
	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 packetlen;					/* Length of packet */
	char data[0];
};	

/* 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[80];		/* Default context for incoming calls */
	char cid_num[80];		/* Caller ID num */
	char cid_name[80];		/* Caller ID name */
	char accountcode[20];		/* Account code */
	char language[MAX_LANGUAGE];	/* Default language for this user */
	char musicclass[MAX_LANGUAGE];  /* Music on Hold class */
	char useragent[256];		/* User agent in SIP request */
	struct ast_codec_pref prefs; /* codec prefs */
	unsigned int callgroup;		/* Call group */
	unsigned int pickupgroup;	/* Pickup Group */
	int flags;			/* SIP_ flags */	
	int nat;			/* NAT setting */
	int amaflags;			/* AMA flags for billing */
	int callingpres;		/* Calling id presentation */
	int insecure;			/* Insecure means don't check password */
	int canreinvite;		/* Do we support re-invites ? */
	int capability;			/* Codec capability */
#ifdef OSP_SUPPORT
	int ospauth;			/* Allow OSP Authentication */
#endif
	int dtmfmode;			/* DTMF setting */
	int inUse;
	int incominglimit;
	int outUse;
	int outgoinglimit;
	int progressinband;
	struct ast_ha *ha;		/* ACL setting */
	struct ast_variable *vars;
};

/* Structure for SIP peer data, we place calls to peers if registred  or fixed IP address (host) */
struct sip_peer {
	ASTOBJ_COMPONENTS(struct sip_peer);
	char secret[80];		/* Password */
	char md5secret[80];		/* Password in MD5 */
	char context[80];		/* Default context for incoming calls */
	char username[80];		/* Temporary username until registration */
	char tohost[80];		/* 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[80];		/* From: domain when calling this peer */
	char fullcontact[128];		/* Contact registred with us (not in sip.conf) */
	char cid_num[80];		/* Caller ID num */
	char cid_name[80];		/* Caller ID name */
	char mailbox[AST_MAX_EXTENSION]; /* Mailbox setting for MWI checks */
	char language[MAX_LANGUAGE];	/* Default language for prompts */
	char musicclass[MAX_LANGUAGE];  /* 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 */
	int flags;			/* SIP_ flags */	
	int expire;			/* Registration expiration */
	int expiry;
	int capability;			/* Codec capability */
	int rtptimeout;
	int rtpholdtimeout;
	int insecure;			/* Do we want to authenticate this peer? */
#ifdef OSP_SUPPORT
	int ospauth;			/* Allow OSP Authentication */
#endif	
	int nat;			/* NAT support needed? */
	int canreinvite;		/* Does the peer support re-invites? */
	unsigned int callgroup;		/* Call group */
	unsigned int pickupgroup;	/* Pickup group */
	int dtmfmode;			/* DTMF mode */
	int progressinband;
	struct sockaddr_in addr;	/* IP address of peer */
	struct in_addr mask;

	/* 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 */
	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

/* NAT settings */
#define SIP_NAT_NEVER		0		/* No nat support */
#define SIP_NAT_RFC3581		(1 << 0)
#define SIP_NAT_ROUTE		(1 << 2)
#define SIP_NAT_ALWAYS		(SIP_NAT_ROUTE | SIP_NAT_RFC3581)

/* 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[80];		/* Domain or host we register to */
	char secret[80];		/* Password or key name in []'s */	
	char md5secret[80];
	char contact[80];		/* Contact extension */
	char random[80];
	int expire;			/* Sched ID of expiration */
	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[256];		/* Authorization realm */
 	char nonce[256];		/* Authorization nonce */
 	char domain[256];		/* Authorization domain */
 	char opaque[256];		/* Opaque nonsense */
 	char qop[80];			/* Quality of Protection. */
 
 	char lastmsg[256];		/* Last Message sent/received */
};

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

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

/*--- 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;


#define REINVITE_INVITE		1
#define REINVITE_UPDATE		2

static int __sip_do_register(struct sip_registry *r);

static int sipsock  = -1;
static int global_nat = SIP_NAT_RFC3581;
static int global_canreinvite = REINVITE_INVITE;


static struct sockaddr_in bindaddr;
static struct sockaddr_in externip;
static struct ast_ha *localaddr;

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_auth(struct sip_pvt *p, char *msg, struct sip_request *req, char *rand, int reliable, char *header);
static int transmit_request(struct sip_pvt *p, char *msg, int inc, int reliable, int newbranch);
static int transmit_request_with_auth(struct sip_pvt *p, char *msg, int inc, int reliable, int newbranch);
static int transmit_invite(struct sip_pvt *p, char *msg, int sendsdp, char *auth, char *authheader, char *vxml_url, char *distinctive_ring, char *osptoken, int addsipheaders, 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_message_with_text(struct sip_pvt *p, char *text);
static int transmit_refer(struct sip_pvt *p, char *dest);
static struct sip_peer *temp_peer(char *name);
static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req, char *header, char *respheader, char *msg, int init);
static void free_old_route(struct sip_route *route);
static int build_reply_digest(struct sip_pvt *p, char *orig_header, char *digest, int digest_len);
static int update_user_counter(struct sip_pvt *fup, int event);
static void prune_peers(void);
static int sip_do_reload(void);


/*--- sip_debug_test_addr: See if we pass debug IP filter */
static inline int sip_debug_test_addr(struct sockaddr_in *addr) 
{
	if (sipdebug == 0)
		return 0;
	if (debugaddr.sin_addr.s_addr) {
		if (((ntohs(debugaddr.sin_port) != 0)
			&& (debugaddr.sin_port != addr->sin_port))
			|| (debugaddr.sin_addr.s_addr != addr->sin_addr.s_addr))
			return 0;
	}
	return 1;
}

static inline int sip_debug_test_pvt(struct sip_pvt *p) 
{
	if (sipdebug == 0)
		return 0;
	return sip_debug_test_addr(((p->nat & SIP_NAT_ROUTE) ? &p->recv : &p->sa));
}


/*--- __sip_xmit: Transmit SIP message ---*/
static int __sip_xmit(struct sip_pvt *p, char *data, int len)
{
	int res;
	char iabuf[INET_ADDRSTRLEN];
	if (p->nat & SIP_NAT_ROUTE)
	    res=sendto(sipsock, data, len, 0, (struct sockaddr *)&p->recv, sizeof(struct sockaddr_in));
	else
	    res=sendto(sipsock, data, len, 0, (struct sockaddr *)&p->sa, sizeof(struct sockaddr_in));
	if (res != len) {
		ast_log(LOG_WARNING, "sip_xmit of %p (len %d) to %s returned %d: %s\n", data, len, ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), res, strerror(errno));
	}
	return res;
}

static void sip_destroy(struct sip_pvt *p);

/*--- ast_sip_ouraddrfor: NAT fix - decide which IP address to use for ASterisk server? ---*/
/* Only used for outbound registrations */
static int ast_sip_ouraddrfor(struct in_addr *them, struct in_addr *us)
{
	/*
	 * Using the localaddr structure built up with localnet statements
	 * apply it to their address to see if we need to substitute our
	 * externip or can get away with our internal bindaddr
	 */
	struct sockaddr_in theirs;
	theirs.sin_addr = *them;
	if (localaddr && externip.sin_addr.s_addr &&
	   ast_apply_ha(localaddr, &theirs)) {
		char iabuf[INET_ADDRSTRLEN];
		memcpy(us, &externip.sin_addr, sizeof(struct in_addr));
		ast_inet_ntoa(iabuf, sizeof(iabuf), *(struct in_addr *)&them->s_addr);
		ast_log(LOG_DEBUG, "Target address %s is not local, substituting externip\n", iabuf);
	}
	else if (bindaddr.sin_addr.s_addr)
		memcpy(us, &bindaddr.sin_addr, sizeof(struct in_addr));
	else
		return ast_ouraddrfor(them, us);
	return 0;
}

static int append_history(struct sip_pvt *p, char *event, char *data)
{
	struct sip_history *hist, *prev;
	char *c;
	if (!recordhistory)
		return 0;
	hist = malloc(sizeof(struct sip_history));
	if (hist) {
		memset(hist, 0, sizeof(struct sip_history));
		snprintf(hist->event, sizeof(hist->event), "%-15s %s", event, data);
		/* Trim up nicely */
		c = hist->event;
		while(*c) {
			if ((*c == '\r') || (*c == '\n')) {
				*c = '\0';
				break;
			}
			c++;
		}
		/* Enqueue into history */
		prev = p->history;
		if (prev) {
			while(prev->next)
				prev = prev->next;
			prev->next = hist;
		} else {
			p->history = hist;
		}
	}
	return 0;
}

/*--- retrans_pkt: Retransmit SIP message if no answer ---*/
static int retrans_pkt(void *data)
{
	struct sip_pkt *pkt=data, *prev, *cur;
	int res = 0;
	char iabuf[INET_ADDRSTRLEN];
	ast_mutex_lock(&pkt->owner->lock);
	if (pkt->retrans < MAX_RETRANS) {
		pkt->retrans++;
		if (sip_debug_test_pvt(pkt->owner)) {
			if (pkt->owner->nat & SIP_NAT_ROUTE)
				ast_verbose("Retransmitting #%d (NAT):\n%s\n to %s:%d\n", pkt->retrans, pkt->data, ast_inet_ntoa(iabuf, sizeof(iabuf), pkt->owner->recv.sin_addr), ntohs(pkt->owner->recv.sin_port));
			else
				ast_verbose("Retransmitting #%d (no NAT):\n%s\n to %s:%d\n", pkt->retrans, pkt->data, ast_inet_ntoa(iabuf, sizeof(iabuf), pkt->owner->sa.sin_addr), ntohs(pkt->owner->sa.sin_port));
		}
		append_history(pkt->owner, "ReTx", pkt->data);
		__sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
		res = 1;
	} else {
		ast_log(LOG_WARNING, "Maximum retries exceeded on call %s for seqno %d (%s %s)\n", pkt->owner->callid, pkt->seqno, (pkt->flags & FLAG_FATAL) ? "Critical" : "Non-critical", (pkt->flags & FLAG_RESPONSE) ? "Response" : "Request");
		append_history(pkt->owner, "MaxRetries", pkt->flags & FLAG_FATAL ? "(Critical)" : "(Non-critical)");
		pkt->retransid = -1;
		if (pkt->flags & FLAG_FATAL) {
			while(pkt->owner->owner && ast_mutex_trylock(&pkt->owner->owner->lock)) {
				ast_mutex_unlock(&pkt->owner->lock);
				usleep(1);
				ast_mutex_lock(&pkt->owner->lock);
			}
			if (pkt->owner->owner) {
				ast_set_flag(pkt->owner, SIP_ALREADYGONE);
				ast_queue_hangup(pkt->owner->owner);
				ast_mutex_unlock(&pkt->owner->owner->lock);
			} else {
				/* If no owner, destroy now */
				ast_set_flag(pkt->owner, SIP_NEEDDESTROY);	
			}
		}
		/* In any case, go ahead and remove the packet */
		prev = NULL;
		cur = pkt->owner->packets;
		while(cur) {
			if (cur == pkt)
				break;
			prev = cur;
			cur = cur->next;
		}
		if (cur) {
			if (prev)
				prev->next = cur->next;
			else
				pkt->owner->packets = cur->next;
			ast_mutex_unlock(&pkt->owner->lock);
			free(cur);
			pkt = NULL;
		} else
			ast_log(LOG_WARNING, "Weird, couldn't find packet owner!\n");
	}
	if (pkt)
		ast_mutex_unlock(&pkt->owner->lock);
	return res;
}

/*--- __sip_reliable_xmit: transmit packet with retransmits ---*/
static int __sip_reliable_xmit(struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal)
{
	struct sip_pkt *pkt;
	pkt = malloc(sizeof(struct sip_pkt) + len + 1);
	if (!pkt)
		return -1;
	memset(pkt, 0, sizeof(struct sip_pkt));
	memcpy(pkt->data, data, len);
	pkt->packetlen = len;
	pkt->next = p->packets;
	pkt->owner = p;
	pkt->seqno = seqno;
	pkt->flags = resp;
	pkt->data[len] = '\0';
	if (fatal)
		pkt->flags |= FLAG_FATAL;
	/* Schedule retransmission */
	pkt->retransid = ast_sched_add(sched, DEFAULT_RETRANS, retrans_pkt, pkt);
	pkt->next = p->packets;
	p->packets = pkt;
	__sip_xmit(pkt->owner, pkt->data, pkt->packetlen);
	if (!strncasecmp(pkt->data, "INVITE", 6)) {
		/* Note this is a pending invite */
		p->pendinginvite = seqno;
	}
	return 0;
}

/*--- __sip_autodestruct: Kill a call (called by scheduler) ---*/
static int __sip_autodestruct(void *data)
{
	struct sip_pvt *p = data;
	p->autokillid = -1;
	ast_log(LOG_DEBUG, "Auto destroying call '%s'\n", p->callid);
	append_history(p, "AutoDestroy", "");
	if (p->owner) {
		ast_log(LOG_WARNING, "Autodestruct on call '%s' with owner in place\n", p->callid);
		ast_queue_hangup(p->owner);
	} else {
		sip_destroy(p);
	}
	return 0;
}

/*--- sip_scheddestroy: Schedule destruction of SIP call ---*/
static int sip_scheddestroy(struct sip_pvt *p, int ms)
{
	char tmp[80];
	if (sip_debug_test_pvt(p))
		ast_verbose("Scheduling destruction of call '%s' in %d ms\n", p->callid, ms);
	if (recordhistory) {
		snprintf(tmp, sizeof(tmp), "%d ms", ms);
		append_history(p, "SchedDestroy", tmp);
	}
	if (p->autokillid > -1)
		ast_sched_del(sched, p->autokillid);
	p->autokillid = ast_sched_add(sched, ms, __sip_autodestruct, p);
	return 0;
}

/*--- sip_cancel_destroy: Cancel destruction of SIP call ---*/
static int sip_cancel_destroy(struct sip_pvt *p)
{
	if (p->autokillid > -1)
		ast_sched_del(sched, p->autokillid);
	append_history(p, "CancelDestroy", "");
	p->autokillid = -1;
	return 0;
}

/*--- __sip_ack: Acknowledges receipt of a packet and stops retransmission ---*/
static int __sip_ack(struct sip_pvt *p, int seqno, int resp, const char *msg)
{
	struct sip_pkt *cur, *prev = NULL;
	int res = -1;
	int resetinvite = 0;
	/* Just in case... */
	if (!msg) msg = "___NEVER___";
	cur = p->packets;
	while(cur) {
		if ((cur->seqno == seqno) && ((cur->flags & FLAG_RESPONSE) == resp) &&
			((cur->flags & FLAG_RESPONSE) || 
			 (!strncasecmp(msg, cur->data, strlen(msg)) && (cur->data[strlen(msg)] < 33)))) {
			if (!resp && (seqno == p->pendinginvite)) {
				ast_log(LOG_DEBUG, "Acked pending invite %d\n", p->pendinginvite);
				p->pendinginvite = 0;
				resetinvite = 1;
			}
			/* this is our baby */
			if (prev)
				prev->next = cur->next;
			else
				p->packets = cur->next;
			if (cur->retransid > -1)
				ast_sched_del(sched, cur->retransid);
			free(cur);
			res = 0;
			break;
		}
		prev = cur;
		cur = cur->next;
	}
	ast_log(LOG_DEBUG, "Stopping retransmission on '%s' of %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found");
	return res;
}

/* Pretend to ack all packets */
static int __sip_pretend_ack(struct sip_pvt *p)
{
	while(p->packets) {
		__sip_ack(p, p->packets->seqno, (p->packets->flags & FLAG_RESPONSE), p->packets->data);
	}
	return 0;
}

/*--- __sip_semi_ack: Acks receipt of packet, keep it around (used for provisional responses) ---*/
static int __sip_semi_ack(struct sip_pvt *p, int seqno, int resp, const char *msg)
{
	struct sip_pkt *cur;
	int res = -1;
	cur = p->packets;
	while(cur) {
		if ((cur->seqno == seqno) && ((cur->flags & FLAG_RESPONSE) == resp) &&
			((cur->flags & FLAG_RESPONSE) || 
			 (!strncasecmp(msg, cur->data, strlen(msg)) && (cur->data[strlen(msg)] < 33)))) {
			/* this is our baby */
			if (cur->retransid > -1)
				ast_sched_del(sched, cur->retransid);
			cur->retransid = -1;
			res = 0;
			break;
		}
		cur = cur->next;
	}
	ast_log(LOG_DEBUG, "(Provisional) Stopping retransmission (but retaining packet) on '%s' %s %d: %s\n", p->callid, resp ? "Response" : "Request", seqno, res ? "Not Found" : "Found");
	return res;
}

static void parse(struct sip_request *req);
static char *get_header(struct sip_request *req, char *name);
static void copy_request(struct sip_request *dst,struct sip_request *src);

static void parse_copy(struct sip_request *dst, struct sip_request *src)
{
	memset(dst, 0, sizeof(*dst));
	memcpy(dst->data, src->data, sizeof(dst->data));
	dst->len = src->len;
	parse(dst);
}
/*--- send_response: Transmit response on SIP request---*/
static int send_response(struct sip_pvt *p, struct sip_request *req, int reliable, int seqno)
{
	int res;
	char iabuf[INET_ADDRSTRLEN];
	struct sip_request tmp;
	char tmpmsg[80];
	if (sip_debug_test_pvt(p)) {
		if (p->nat & SIP_NAT_ROUTE)
			ast_verbose("%sTransmitting (NAT):\n%s\n to %s:%d\n", reliable ? "Reliably " : "", req->data, ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port));
		else
			ast_verbose("%sTransmitting (no NAT):\n%s\n to %s:%d\n", reliable ? "Reliably " : "", req->data, ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port));
	}
	if (reliable) {
		if (recordhistory) {
			parse_copy(&tmp, req);
			snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq"));
			append_history(p, "TxRespRel", tmpmsg);
		}
		res = __sip_reliable_xmit(p, seqno, 1, req->data, req->len, (reliable > 1));
	} else {
		if (recordhistory) {
			parse_copy(&tmp, req);
			snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq"));
			append_history(p, "TxResp", tmpmsg);
		}
		res = __sip_xmit(p, req->data, req->len);
	}
	if (res > 0)
		res = 0;
	return res;
}

/*--- send_request: Send SIP Request to the other part of the dialogue ---*/
static int send_request(struct sip_pvt *p, struct sip_request *req, int reliable, int seqno)
{
	int res;
	char iabuf[INET_ADDRSTRLEN];
	struct sip_request tmp;
	char tmpmsg[80];
	if (sip_debug_test_pvt(p)) {
		if (p->nat & SIP_NAT_ROUTE)
			ast_verbose("%sTransmitting:\n%s (NAT) to %s:%d\n", reliable ? "Reliably " : "", req->data, ast_inet_ntoa(iabuf, sizeof(iabuf), p->recv.sin_addr), ntohs(p->recv.sin_port));
		else
			ast_verbose("%sTransmitting:\n%s (no NAT) to %s:%d\n", reliable ? "Reliably " : "", req->data, ast_inet_ntoa(iabuf, sizeof(iabuf), p->sa.sin_addr), ntohs(p->sa.sin_port));
	}
	if (reliable) {
		if (recordhistory) {
			parse_copy(&tmp, req);
			snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq"));
			append_history(p, "TxReqRel", tmpmsg);
		}
		res = __sip_reliable_xmit(p, seqno, 0, req->data, req->len, (reliable > 1));
	} else {
		if (recordhistory) {
			parse_copy(&tmp, req);
			snprintf(tmpmsg, sizeof(tmpmsg), "%s / %s", tmp.data, get_header(&tmp, "CSeq"));
			append_history(p, "TxReq", tmpmsg);
		}
		res = __sip_xmit(p, req->data, req->len);
	}
	return res;
}

/*--- url_decode: Decode SIP URL  ---*/
static void url_decode(char *s) 
{
	char *o = s;
	unsigned int tmp;
	while(*s) {
		switch(*s) {
		case '%':
			if (strlen(s) > 2) {
				if (sscanf(s + 1, "%2x", &tmp) == 1) {
					*o = tmp;
					s += 2;	/* Will be incremented once more when we break out */
					break;
				}
			}
			/* Fall through if something wasn't right with the formatting */
		default:
			*o = *s;
		}
		s++;
		o++;
	}
	*o = '\0';
}

/*--- ditch_braces: Pick out text in braces from character string  ---*/
static char *ditch_braces(char *tmp)
{
	char *c = tmp;
	char *n;
	char *q;
	if ((q = strchr(tmp, '"')) ) {
		c = q + 1;
		if ((q = strchr(c, '"')) )
			c = q + 1;
		else {
			ast_log(LOG_WARNING, "No closing quote in '%s'\n", tmp);
			c = tmp;
		}
	}