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Asterisk developer's documentation

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chan_sip.c

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/*
 * Asterisk -- An open source telephony toolkit.
 *
 * Copyright (C) 1999 - 2006, 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
 *
 * \author Mark Spencer <markster@digium.com>
 *
 * 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: 153114 $")

#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 SIP_RESERVED ";/?:@&=+$,# "

/* #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
   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 xmittype {
   XMIT_CRITICAL = 2,              /*!< Transmit critical SIP message reliably, with re-transmits.
                                              If it fails, it's critical and will cause a teardown of the session */
   XMIT_RELIABLE = 1,              /*!< Transmit SIP message reliably, with re-transmits */
   XMIT_UNRELIABLE = 0,            /*!< Transmit SIP message without bothering with re-transmits */
};

enum parse_register_result {
   PARSE_REGISTER_FAILED,
   PARSE_REGISTER_UPDATE,
   PARSE_REGISTER_QUERY,
};

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

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" },
   /* RFC 4235: SIP Dialog event package */
   { 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 */
   { MWI_NOTIFICATION,  "message-summary", "application/simple-message-summary", "mwi" } /* RFC 3842: Mailbox notification */
};

/*! \brief SIP Request methods known by Asterisk */
enum sipmethod {
   SIP_UNKNOWN,      /* Unknown response */
   SIP_RESPONSE,     /* Not request, response to outbound request */
   SIP_REGISTER,
   SIP_OPTIONS,
   SIP_NOTIFY,
   SIP_INVITE,
   SIP_ACK,
   SIP_PRACK,     /* Not supported at all */
   SIP_BYE,
   SIP_REFER,
   SIP_SUBSCRIBE,
   SIP_MESSAGE,
   SIP_UPDATE,    /* We can send UPDATE; but not accept it */
   SIP_INFO,
   SIP_CANCEL,
   SIP_PUBLISH,      /* Not supported at all */
   SIP_PING,      /* Not supported at all, no standard but still implemented out there */
};

/*! \brief Authentication types - proxy or www authentication 
   \note Endpoints, like Asterisk, should always use WWW authentication to
   allow multiple authentications in the same call - to the proxy and
   to the end point.
*/
enum sip_auth_type {
   PROXY_AUTH,
   WWW_AUTH,
};

/*! \brief Authentication result from check_auth* functions */
enum check_auth_result {
   AUTH_SUCCESSFUL = 0,
   AUTH_CHALLENGE_SENT = 1,
   AUTH_SECRET_FAILED = -1,
   AUTH_USERNAME_MISMATCH = -2,
   AUTH_NOT_FOUND = -3,
   AUTH_FAKE_AUTH = -4,
   AUTH_UNKNOWN_DOMAIN = -5,
   AUTH_PEER_NOT_DYNAMIC = -6,
   AUTH_ACL_FAILED = -7,
};

/*! \brief States for outbound registrations (with register= lines in sip.conf */
enum sipregistrystate {
   REG_STATE_UNREGISTERED = 0,   /*!< We are not registred */
   REG_STATE_REGSENT,   /*!< Registration request sent */
   REG_STATE_AUTHSENT,  /*!< We have tried to authenticate */
   REG_STATE_REGISTERED,   /*!< Registred and done */
   REG_STATE_REJECTED,  /*!< Registration rejected */
   REG_STATE_TIMEOUT,   /*!< Registration timed out */
   REG_STATE_NOAUTH, /*!< We have no accepted credentials */
   REG_STATE_FAILED, /*!< Registration failed after several tries */
};

#define CAN_NOT_CREATE_DIALOG 0
#define CAN_CREATE_DIALOG  1
#define CAN_CREATE_DIALOG_UNSUPPORTED_METHOD 2

/*! XXX Note that sip_methods[i].id == i must hold or the code breaks */
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;
   int can_create;
} sip_methods[] = {
   { SIP_UNKNOWN,  RTP,    "-UNKNOWN-",   CAN_CREATE_DIALOG },
   { SIP_RESPONSE,    NO_RTP, "SIP/2.0",  CAN_NOT_CREATE_DIALOG },
   { SIP_REGISTER,    NO_RTP, "REGISTER",    CAN_CREATE_DIALOG },
   { SIP_OPTIONS,  NO_RTP, "OPTIONS",  CAN_CREATE_DIALOG },
   { SIP_NOTIFY,   NO_RTP, "NOTIFY",   CAN_CREATE_DIALOG },
   { SIP_INVITE,   RTP,    "INVITE",   CAN_CREATE_DIALOG },
   { SIP_ACK,   NO_RTP, "ACK",   CAN_NOT_CREATE_DIALOG },
   { SIP_PRACK,    NO_RTP, "PRACK",    CAN_NOT_CREATE_DIALOG },
   { SIP_BYE,   NO_RTP, "BYE",   CAN_NOT_CREATE_DIALOG },
   { SIP_REFER,    NO_RTP, "REFER",    CAN_CREATE_DIALOG },
   { SIP_SUBSCRIBE, NO_RTP, "SUBSCRIBE",  CAN_CREATE_DIALOG },
   { SIP_MESSAGE,  NO_RTP, "MESSAGE",  CAN_CREATE_DIALOG },
   { SIP_UPDATE,   NO_RTP, "UPDATE",   CAN_NOT_CREATE_DIALOG },
   { SIP_INFO,  NO_RTP, "INFO",  CAN_NOT_CREATE_DIALOG },
   { SIP_CANCEL,   NO_RTP, "CANCEL",   CAN_NOT_CREATE_DIALOG },
   { SIP_PUBLISH,  NO_RTP, "PUBLISH",  CAN_CREATE_DIALOG_UNSUPPORTED_METHOD },
   { SIP_PING,  NO_RTP, "PING",  CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }
};

/*!  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)
#define SIP_OPT_NOREFERSUB (1 << 14)
#define SIP_OPT_HISTINFO   (1 << 15)
#define SIP_OPT_RESPRIORITY   (1 << 16)

/*! \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[] = {  /* XXX used in 3 places */
   /* RFC3891: Replaces: header for transfer */
   { SIP_OPT_REPLACES,  SUPPORTED,  "replaces" },  
   /* One version of Polycom firmware has the wrong label */
   { SIP_OPT_REPLACES,  SUPPORTED,  "replace" },   
   /* RFC3262: PRACK 100% reliability */
   { SIP_OPT_100REL, NOT_SUPPORTED, "100rel" }, 
   /* RFC4028: SIP Session Timers */
   { SIP_OPT_TIMER,  NOT_SUPPORTED, "timer" },
   /* RFC3959: SIP Early session support */
   { SIP_OPT_EARLY_SESSION, NOT_SUPPORTED,   "early-session" },
   /* RFC3911: 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:  RFC4662 */
   { SIP_OPT_EVENTLIST, NOT_SUPPORTED, "eventlist" },
   /* GRUU: Globally Routable User Agent URI's */
   { SIP_OPT_GRUU,      NOT_SUPPORTED, "gruu" },
   /* RFC4538: Target-dialog */
   { SIP_OPT_TARGET_DIALOG,NOT_SUPPORTED, "tdialog" },
   /* Disable the REFER subscription, RFC 4488 */
   { SIP_OPT_NOREFERSUB,   NOT_SUPPORTED, "norefersub" },
   /* ietf-sip-history-info-06.txt */
   { SIP_OPT_HISTINFO,  NOT_SUPPORTED, "histinfo" },
   /* ietf-sip-resource-priority-10.txt */
   { SIP_OPT_RESPRIORITY,  NOT_SUPPORTED, "resource-priority" },
};


/*! \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" 

/*! \brief Standard SIP port from RFC 3261. DO NOT CHANGE THIS */
#define STANDARD_SIP_PORT  5060
/* Note: in many SIP headers, absence of a port number implies port 5060,
 * and this is why we cannot change the above constant.
 * There is a limited number of places in asterisk where we could,
 * in principle, use a different "default" port number, but
 * we do not support this feature at the moment.
 */

/* Default values, set and reset in reload_config before reading configuration */
/* These are default values in the source. There are other recommended values in the
   sip.conf.sample for new installations. These may differ to keep backwards compatibility,
   yet encouraging new behaviour on new installations 
 */
#define DEFAULT_CONTEXT    "default"
#define DEFAULT_MOHINTERPRET    "default"
#define DEFAULT_MOHSUGGEST      ""
#define DEFAULT_VMEXTEN    "asterisk"
#define DEFAULT_CALLERID   "asterisk"
#define DEFAULT_NOTIFYMIME    "application/simple-message-summary"
#define DEFAULT_MWITIME    10
#define DEFAULT_ALLOWGUEST TRUE
#define DEFAULT_SRVLOOKUP  TRUE     /*!< Recommended setting is ON */
#define DEFAULT_COMPACTHEADERS   FALSE
#define DEFAULT_TOS_SIP         0               /*!< Call signalling packets should be marked as DSCP CS3, but the default is 0 to be compatible with previous versions. */
#define DEFAULT_TOS_AUDIO       0               /*!< Audio packets should be marked as DSCP EF (Expedited Forwarding), but the default is 0 to be compatible with previous versions. */
#define DEFAULT_TOS_VIDEO       0               /*!< Video packets should be marked as DSCP AF41, but the default is 0 to be compatible with previous versions. */
#define DEFAULT_ALLOW_EXT_DOM TRUE
#define DEFAULT_REALM      "asterisk"
#define DEFAULT_NOTIFYRINGING TRUE
#define DEFAULT_PEDANTIC   FALSE
#define DEFAULT_AUTOCREATEPEER   FALSE
#define DEFAULT_QUALIFY    FALSE
#define DEFAULT_T1MIN      100      /*!< 100 MS for minimal roundtrip time */
#define DEFAULT_MAX_CALL_BITRATE (384)    /*!< Max bitrate for video */
#ifndef DEFAULT_USERAGENT
#define DEFAULT_USERAGENT "Asterisk PBX"  /*!< Default Useragent: header unless re-defined in sip.conf */
#endif


/* Default setttings are used as a channel setting and as a default when
   configuring devices */
static char default_context[AST_MAX_CONTEXT];
static char default_subscribecontext[AST_MAX_CONTEXT];
static char default_language[MAX_LANGUAGE];
static char default_callerid[AST_MAX_EXTENSION];
static char default_fromdomain[AST_MAX_EXTENSION];
static char default_notifymime[AST_MAX_EXTENSION];
static int default_qualify;      /*!< Default Qualify= setting */
static char default_vmexten[AST_MAX_EXTENSION];
static char default_mohinterpret[MAX_MUSICCLASS];  /*!< Global setting for moh class to use when put on hold */
static char default_mohsuggest[MAX_MUSICCLASS];    /*!< Global setting for moh class to suggest when putting 
                                                    *   a bridged channel on hold */
static int default_maxcallbitrate;  /*!< Maximum bitrate for call */
static struct ast_codec_pref default_prefs;     /*!< Default codec prefs */

/* Global settings only apply to the channel */
static int global_directrtpsetup;   /*!< Enable support for Direct RTP setup (no re-invites) */
static int global_limitonpeers;     /*!< Match call limit on peers only */
static int global_rtautoclear;
static int global_notifyringing; /*!< Send notifications on ringing */
static int global_notifyhold;    /*!< Send notifications on hold */
static int global_alwaysauthreject; /*!< Send 401 Unauthorized for all failing requests */
static int srvlookup;         /*!< SRV Lookup on or off. Default is on */
static int pedanticsipchecking;     /*!< Extra checking ?  Default off */
static int autocreatepeer;    /*!< Auto creation of peers at registration? Default off. */
static int global_relaxdtmf;        /*!< Relax DTMF */
static int global_rtptimeout;    /*!< Time out call if no RTP */
static int global_rtpholdtimeout;
static int global_rtpkeepalive;     /*!< Send RTP keepalives */
static int global_reg_timeout;   
static int global_regattempts_max;  /*!< Registration attempts before giving up */
static int global_allowguest;    /*!< allow unauthenticated users/peers to connect? */
static int global_allowsubscribe;   /*!< Flag for disabling ALL subscriptions, this is FALSE only if all peers are FALSE 
                   the global setting is in globals_flags[1] */
static int global_mwitime;    /*!< Time between MWI checks for peers */
static unsigned int global_tos_sip;    /*!< IP type of service for SIP packets */
static unsigned int global_tos_audio;     /*!< IP type of service for audio RTP packets */
static unsigned int global_tos_video;     /*!< IP type of service for video RTP packets */
static int compactheaders;    /*!< send compact sip headers */
static int recordhistory;     /*!< Record SIP history. Off by default */
static int dumphistory;       /*!< Dump history to verbose before destroying SIP dialog */
static char global_realm[MAXHOSTNAMELEN];       /*!< Default realm */
static char global_regcontext[AST_MAX_CONTEXT];    /*!< Context for auto-extensions */
static char global_useragent[AST_MAX_EXTENSION];   /*!< Useragent for the SIP channel */
static int allow_external_domains;  /*!< Accept calls to external SIP domains? */
static int global_callevents;    /*!< Whether we send manager events or not */
static int global_t1min;      /*!< T1 roundtrip time minimum */
static int global_autoframing;          /*!< Turn autoframing on or off. */
static enum transfermodes global_allowtransfer; /*!< SIP Refer restriction scheme */

static int global_matchexterniplocally; /*!< Match externip/externhost setting against localnet setting */

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

/*! \brief Global list of addresses dynamic peers are not allowed to use */
static struct ast_ha *global_contact_ha = NULL;
static int global_dynamic_exclude_static = 0;

/* Object counters */
static int suserobjs = 0;                /*!< Static users */
static int ruserobjs = 0;                /*!< Realtime users */
static int speerobjs = 0;                /*!< Statis peers */
static int rpeerobjs = 0;                /*!< Realtime peers */
static int apeerobjs = 0;                /*!< Autocreated peer objects */
static int regobjs = 0;                  /*!< Registry objects */

static struct ast_flags global_flags[2] = {{0}};        /*!< global SIP_ flags */

/*! \brief Protect the SIP dialog 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);

AST_MUTEX_DEFINE_STATIC(sip_reload_lock);

/*! \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 sip_reloading = FALSE;                       /*!< Flag for avoiding multiple reloads at the same time */
static enum channelreloadreason sip_reloadreason;       /*!< Reason for last reload/load of configuration */

static struct sched_context *sched;     /*!< The scheduling context */
static struct io_context *io;           /*!< The IO context */
static int *sipsock_read_id;            /*!< ID of IO entry for sipsock FD */

#define DEC_CALL_LIMIT  0
#define INC_CALL_LIMIT  1
#define DEC_CALL_RINGING 2
#define INC_CALL_RINGING 3

/*! \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 */
   int lines;              /*!< Body Content */
   unsigned int flags;     /*!< SIP_PKT Flags for this packet */
   char *header[SIP_MAX_HEADERS];
   char *line[SIP_MAX_LINES];
   char data[SIP_MAX_PACKET];
   unsigned int sdp_start; /*!< the line number where the SDP begins */
   unsigned int sdp_end;   /*!< the line number where the SDP ends */
};

/*
 * A sip packet is stored into the data[] buffer, with the header followed
 * by an empty line and the body of the message.
 * On outgoing packets, data is accumulated in data[] with len reflecting
 * the next available byte, headers and lines count the number of lines
 * in both parts. There are no '\0' in data[0..len-1].
 *
 * On received packet, the input read from the socket is copied into data[],
 * len is set and the string is NUL-terminated. Then a parser fills up
 * the other fields -header[] and line[] to point to the lines of the
 * message, rlPart1 and rlPart2 parse the first lnie as below:
 *
 * Requests have in the first line  METHOD URI SIP/2.0
 * rlPart1 = method; rlPart2 = uri;
 * Responses have in the first line SIP/2.0 code description
 * rlPart1 = SIP/2.0; rlPart2 = code + description;
 *
 */

/*! \brief structure used in transfers */
struct sip_dual {
   struct ast_channel *chan1; /*!< First channel involved */
   struct ast_channel *chan2; /*!< Second channel involved */
   struct sip_request req;    /*!< Request that caused the transfer (REFER) */
   int seqno;        /*!< Sequence number */
};

struct sip_pkt;

/*! \brief Parameters to the transmit_invite function */
struct sip_invite_param {
   const char *distinctive_ring; /*!< Distinctive ring header */
   int addsipheaders;      /*!< Add extra SIP headers */
   const char *uri_options;   /*!< URI options to add to the URI */
   const char *vxml_url;      /*!< VXML url for Cisco phones */
   char *auth;       /*!< Authentication */
   char *authheader;    /*!< Auth header */
   enum sip_auth_type auth_type; /*!< Authentication type */
   const char *replaces;      /*!< Replaces header for call transfers */
   int transfer;        /*!< Flag - is this Invite part of a SIP transfer? (invite/replaces) */
};

/*! \brief Structure to save routing information for a SIP session */
struct sip_route {
   struct sip_route *next;
   char hop[0];
};

/*! \brief Modes for SIP domain handling in the PBX */
enum domain_mode {
   SIP_DOMAIN_AUTO,     /*!< This domain is auto-configured */
   SIP_DOMAIN_CONFIG,      /*!< This domain is from configuration */
};

/*! \brief Domain data structure. 
   \note In the future, we will connect this to a configuration tree specific
   for this domain
*/
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 */


/*! \brief sip_history: Structure for saving transactions within a SIP dialog */
struct sip_history {
   AST_LIST_ENTRY(sip_history) list;
   char event[0]; /* actually more, depending on needs */
};

AST_LIST_HEAD_NOLOCK(sip_history_head, sip_history); /*!< history list, entry in sip_pvt */

/*! \brief sip_auth: Credentials 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 */
};

/*--- Various flags for the flags field in the pvt structure */
#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 by the monitor thread */
#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)   /*!< Direction of the last transaction in this dialog */
#define SIP_FREE_BIT    (1 << 14)   /*!< ---- */
#define SIP_DEFER_BYE_ON_TRANSFER   (1 << 15)   /*!< Do not hangup at first ast_hangup */
#define SIP_DTMF     (3 <<