chan_sip.c

Asterisk中信道部分的源码 。。。。

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;

/* 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 << 16)	/*!< DTMF Support: four settings, uses two bits */
#define SIP_DTMF_RFC2833	(0 << 16)	/*!< DTMF Support: RTP DTMF - "rfc2833" */
#define SIP_DTMF_INBAND		(1 << 16)	/*!< DTMF Support: Inband audio, only for ULAW/ALAW - "inband" */
#define SIP_DTMF_INFO		(2 << 16)	/*!< DTMF Support: SIP Info messages - "info" */
#define SIP_DTMF_AUTO		(3 << 16)	/*!< DTMF Support: 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)	/*!< NAT RFC3581 */
#define SIP_NAT_ROUTE		(2 << 18)	/*!< NAT Only ROUTE */
#define SIP_NAT_ALWAYS		(3 << 18)	/*!< NAT Both ROUTE and RFC3581 */
/* re-INVITE related settings */
#define SIP_REINVITE		(7 << 20)	/*!< three bits used */
#define SIP_CAN_REINVITE	(1 << 20)	/*!< allow peers to be reinvited to send media directly p2p */
#define SIP_CAN_REINVITE_NAT	(2 << 20)	/*!< allow media reinvite when new peer is behind NAT */
#define SIP_REINVITE_UPDATE	(4 << 20)	/*!< use UPDATE (RFC3311) when reinviting this peer */
/* "insecure" settings */
#define SIP_INSECURE_PORT	(1 << 23)	/*!< don't require matching port for incoming requests */
#define SIP_INSECURE_INVITE	(1 << 24)	/*!< don't require authentication for incoming INVITEs */
/* Sending PROGRESS in-band settings */
#define SIP_PROG_INBAND		(3 << 25)	/*!< three settings, uses two bits */
#define SIP_PROG_INBAND_NEVER	(0 << 25)
#define SIP_PROG_INBAND_NO	(1 << 25)
#define SIP_PROG_INBAND_YES	(2 << 25)
#define SIP_NO_HISTORY		(1 << 27)	/*!< Suppress recording request/response history */
#define SIP_CALL_LIMIT		(1 << 28)	/*!< Call limit enforced for this call */
#define SIP_SENDRPID		(1 << 29)	/*!< Remote Party-ID Support */
#define SIP_INC_COUNT		(1 << 30)	/*!< Did this connection increment the counter of in-use calls? */
#define SIP_G726_NONSTANDARD	(1 << 31)	/*!< Use non-standard packing for G726-32 data */

#define SIP_FLAGS_TO_COPY \
	(SIP_PROMISCREDIR | SIP_TRUSTRPID | SIP_SENDRPID | SIP_DTMF | SIP_REINVITE | \
	 SIP_PROG_INBAND | SIP_USECLIENTCODE | SIP_NAT | SIP_G726_NONSTANDARD | \
	 SIP_USEREQPHONE | SIP_INSECURE_PORT | SIP_INSECURE_INVITE)

/*--- a new page of flags (for flags[1] */
/* realtime flags */
#define SIP_PAGE2_RTCACHEFRIENDS	(1 << 0)
#define SIP_PAGE2_RTUPDATE		(1 << 1)
#define SIP_PAGE2_RTAUTOCLEAR		(1 << 2)
#define SIP_PAGE2_RT_FROMCONTACT 	(1 << 4)
#define SIP_PAGE2_RTSAVE_SYSNAME 	(1 << 5)
/* Space for addition of other realtime flags in the future */
#define SIP_PAGE2_STATECHANGEQUEUE	(1 << 9)	/*!< D: Unsent state pending change exists */
#define SIP_PAGE2_IGNOREREGEXPIRE	(1 << 10)
#define SIP_PAGE2_DEBUG			(3 << 11)
#define SIP_PAGE2_DEBUG_CONFIG 		(1 << 11)
#define SIP_PAGE2_DEBUG_CONSOLE 	(1 << 12)
#define SIP_PAGE2_DYNAMIC		(1 << 13)	/*!< Dynamic Peers register with Asterisk */
#define SIP_PAGE2_SELFDESTRUCT		(1 << 14)	/*!< Automatic peers need to destruct themselves */
#define SIP_PAGE2_VIDEOSUPPORT		(1 << 15)
#define SIP_PAGE2_ALLOWSUBSCRIBE	(1 << 16)	/*!< Allow subscriptions from this peer? */
#define SIP_PAGE2_ALLOWOVERLAP		(1 << 17)	/*!< Allow overlap dialing ? */
#define SIP_PAGE2_SUBSCRIBEMWIONLY	(1 << 18)	/*!< Only issue MWI notification if subscribed to */
#define SIP_PAGE2_INC_RINGING		(1 << 19)	/*!< Did this connection increment the counter of in-use calls? */
#define SIP_PAGE2_T38SUPPORT		(7 << 20)	/*!< T38 Fax Passthrough Support */
#define SIP_PAGE2_T38SUPPORT_UDPTL	(1 << 20)	/*!< 20: T38 Fax Passthrough Support */
#define SIP_PAGE2_T38SUPPORT_RTP	(2 << 20)	/*!< 21: T38 Fax Passthrough Support (not implemented) */
#define SIP_PAGE2_T38SUPPORT_TCP	(4 << 20)	/*!< 22: T38 Fax Passthrough Support (not implemented) */
#define SIP_PAGE2_CALL_ONHOLD		(3 << 23)	/*!< Call states */
#define SIP_PAGE2_CALL_ONHOLD_ACTIVE    (1 << 23)       /*!< 23: Active hold */
#define SIP_PAGE2_CALL_ONHOLD_ONEDIR	(2 << 23)	/*!< 23: One directional hold */
#define SIP_PAGE2_CALL_ONHOLD_INACTIVE	(3 << 23)	/*!< 23: Inactive hold */
#define SIP_PAGE2_RFC2833_COMPENSATE    (1 << 25)	/*!< 25: ???? */
#define SIP_PAGE2_BUGGY_MWI		(1 << 26)	/*!< 26: Buggy CISCO MWI fix */
#define SIP_PAGE2_OUTGOING_CALL         (1 << 27)       /*!< 27: Is this an outgoing call? */
#define SIP_PAGE2_UDPTL_DESTINATION     (1 << 28)       /*!< 28: Use source IP of RTP as destination if NAT is enabled */
#define SIP_PAGE2_DIALOG_ESTABLISHED    (1 << 29)       /*!< 29: Has a dialog been established? */

#define SIP_PAGE2_FLAGS_TO_COPY \
	(SIP_PAGE2_ALLOWSUBSCRIBE | SIP_PAGE2_ALLOWOVERLAP | SIP_PAGE2_VIDEOSUPPORT | \
	SIP_PAGE2_T38SUPPORT | SIP_PAGE2_RFC2833_COMPENSATE | SIP_PAGE2_BUGGY_MWI | SIP_PAGE2_UDPTL_DESTINATION)

/* 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 */