chan_sip.c File Reference

Detailed Description

Implementation of Session Initiation Protocol.

Author:: Mark Spencer <markster@digium.com>

See Also:

CREDITS

Implementation of RFC 3261 - without S/MIME, TCP and TLS support Configuration file sip.conf

Todo:

SIP over TCP

SIP over TLS

Better support of forking

VIA branch tag transaction checking

Transaction support

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"

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

Outbound calls: Outbound calls are set up by the PBX through the sip_request_call() function. After that, they are activated by sip_call().

Hanging up: The PBX issues a hangup on both incoming and outgoing calls through the sip_hangup() function

Definition in file chan_sip.c.

#include "asterisk.h"
#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/dnsmgr.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"
#include "asterisk/version.h"

Include dependency graph for chan_sip.c:

Go to the source code of this file.

Data Structures

struct ast_peer_list

The peer list: Peers and Friends. More...

struct ast_register_list

The register list: Other SIP proxies we register with and place calls to. More...

struct ast_user_list

The user list: Users and friends. More...

struct c_referstatusstring

Table to convert from REFER status variable to string. More...

struct cfsip_methods

Structure for parsing of SIP methods. More...

struct cfsip_options

List of well-known SIP options. If we get this in a require, we should check the list and answer accordingly. More...

struct cfsubscription_types

Description of SUBSCRIBE events. More...

struct domain

Domain data structure. More...

struct sip_auth

sip_auth: Credentials for authentication to other SIP services More...

struct sip_dual

structure used in transfers More...

struct sip_history

sip_history: Structure for saving transactions within a SIP dialog More...

struct sip_invite_param

Parameters to the transmit_invite function. More...

struct sip_peer

Structure for SIP peer data, we place calls to peers if registered or fixed IP address (host). More...

struct sip_pkt

sip packet - raw format for outbound packets that are sent or scheduled for transmission More...

struct sip_pvt

sip_pvt: PVT structures are used for each SIP dialog, ie. a call, a registration, a subscribe More...

struct sip_refer

Structure to handle SIP transfers. Dynamically allocated when needed. More...

struct sip_registry

Registrations with other SIP proxies. More...

struct sip_request

sip_request: The data grabbed from the UDP socket More...

struct sip_route

Structure to save routing information for a SIP session. More...

struct sip_user

Structure for SIP user data. User's place calls to us. More...

struct t38properties

T.38 channel settings (at some point we need to make this alloc'ed. More...

Defines

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

SIP Methods we support.

#define append_history(p, event, fmt, args...)   append_history_full(p, "%-15s " fmt, event, ## args)

Append to SIP dialog history.

#define CALLERID_UNKNOWN   "Unknown"

#define DEC_CALL_LIMIT   0

#define DEC_CALL_RINGING   2

#define DEFAULT_ALLOW_EXT_DOM   TRUE

#define DEFAULT_ALLOWGUEST   TRUE

#define DEFAULT_AUTOCREATEPEER   FALSE

#define DEFAULT_CALLERID   "asterisk"

#define DEFAULT_COMPACTHEADERS   FALSE

#define DEFAULT_CONTEXT   "default"

#define DEFAULT_DEFAULT_EXPIRY   120

#define DEFAULT_EXPIRY   900

#define DEFAULT_FREQ_NOTOK   10 * 1000

#define DEFAULT_FREQ_OK   60 * 1000

#define DEFAULT_MAX_CALL_BITRATE   (384)

#define DEFAULT_MAX_EXPIRY   3600

#define DEFAULT_MAX_FORWARDS   "70"

#define DEFAULT_MAXMS   2000

#define DEFAULT_MIN_EXPIRY   60

#define DEFAULT_MOHINTERPRET   "default"

#define DEFAULT_MOHSUGGEST   ""

#define DEFAULT_MWITIME   10

#define DEFAULT_NOTIFYMIME   "application/simple-message-summary"

#define DEFAULT_NOTIFYRINGING   TRUE

#define DEFAULT_PEDANTIC   FALSE

#define DEFAULT_QUALIFY   FALSE

#define DEFAULT_REALM   "asterisk"

#define DEFAULT_REGISTRATION_TIMEOUT   20

#define DEFAULT_RETRANS   1000

#define DEFAULT_SRVLOOKUP   FALSE

#define DEFAULT_T1MIN   100

#define DEFAULT_TOS_AUDIO   0

#define DEFAULT_TOS_SIP   0

#define DEFAULT_TOS_VIDEO   0

#define DEFAULT_TRANS_TIMEOUT   -1

#define DEFAULT_USERAGENT   "Asterisk PBX"

#define DEFAULT_VMEXTEN   "asterisk"

#define EXPIRY_GUARD_LIMIT   30

#define EXPIRY_GUARD_MIN   500

#define EXPIRY_GUARD_PCT   0.20

#define EXPIRY_GUARD_SECS   15

#define FALSE   0

#define FLAG_FATAL   (1 << 1)

#define FLAG_RESPONSE   (1 << 0)

#define FORMAT   "%-15.15s %-10.10s %-11.11s %5.5d/%5.5d %-4.4s %-3.3s %-3.3s %-15.15s %-10.10s\n"

#define FORMAT   "%-30.30s %-12.12s %8d %-20.20s %-25.25s\n"

#define FORMAT   "%-40.40s %-20.20s %-16.16s\n"

#define FORMAT   "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8d %-10s %-10s\n"

#define FORMAT   "%-25.25s %-15.15s %-15.15s %-15.15s %-5.5s%-10.10s\n"

#define FORMAT   "%-25.25s %-15.15s %-15.15s \n"

#define FORMAT2   "%-15.15s %-10.10s %-11.11s %-11.11s %-4.4s %-7.7s %-15.15s\n"

#define FORMAT2   "%-30.30s %-12.12s %8.8s %-20.20s %-25.25s\n"

#define FORMAT2   "%-25.25s %-15.15s %-3.3s %-3.3s %-3.3s %-8s %-10s %-10s\n"

#define FORMAT2   "%-25.25s %-15.15s %-15.15s \n"

#define FORMAT3   "%-15.15s %-10.10s %-11.11s %-15.15s %-13.13s %-15.15s %-10.10s\n"

#define INC_CALL_LIMIT   1

#define INC_CALL_RINGING   3

#define INITIAL_CSEQ   101

#define IPTOS_MINCOST   0x02

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

#define MAX_AUTHTRIES   3

#define MAX_RETRANS   6

#define NO_RTP   0

#define NOT_SUPPORTED   0

#define RTP   1

#define SDP_SAMPLE_RATE(x)   (x == AST_FORMAT_G722) ? 16000 : 8000

#define SIP_ALREADYGONE   (1 << 0)

#define SIP_CALL_LIMIT   (1 << 28)

#define SIP_CAN_REINVITE   (1 << 20)

#define SIP_CAN_REINVITE_NAT   (2 << 20)

#define SIP_DEFER_BYE_ON_TRANSFER   (1 << 15)

#define SIP_DTMF   (3 << 16)

#define SIP_DTMF_AUTO   (3 << 16)

#define SIP_DTMF_INBAND   (1 << 16)

#define SIP_DTMF_INFO   (2 << 16)

#define SIP_DTMF_RFC2833   (0 << 16)

#define SIP_FLAGS_TO_COPY

#define SIP_FREE_BIT   (1 << 14)

#define SIP_G726_NONSTANDARD   (1 << 31)

#define SIP_GOTREFER   (1 << 7)

#define SIP_INC_COUNT   (1 << 30)

#define SIP_INSECURE_INVITE   (1 << 24)

#define SIP_INSECURE_PORT   (1 << 23)

#define SIP_MAX_HEADERS   64

#define SIP_MAX_LINES   64

#define SIP_MAX_PACKET   4096

#define SIP_NAT   (3 << 18)

#define SIP_NAT_ALWAYS   (3 << 18)

#define SIP_NAT_NEVER   (0 << 18)

#define SIP_NAT_RFC3581   (1 << 18)

#define SIP_NAT_ROUTE   (2 << 18)

#define SIP_NEEDDESTROY   (1 << 1)

#define SIP_NEEDREINVITE   (1 << 5)

#define SIP_NO_HISTORY   (1 << 27)

#define SIP_NOVIDEO   (1 << 2)

#define SIP_OPT_100REL   (1 << 1)

#define SIP_OPT_EARLY_SESSION   (1 << 3)

#define SIP_OPT_EVENTLIST   (1 << 11)

#define SIP_OPT_GRUU   (1 << 12)

#define SIP_OPT_HISTINFO   (1 << 15)

#define SIP_OPT_JOIN   (1 << 4)

#define SIP_OPT_NOREFERSUB   (1 << 14)

#define SIP_OPT_PATH   (1 << 5)

#define SIP_OPT_PRECONDITION   (1 << 7)

#define SIP_OPT_PREF   (1 << 6)

#define SIP_OPT_PRIVACY   (1 << 8)

#define SIP_OPT_REPLACES   (1 << 0)

#define SIP_OPT_RESPRIORITY   (1 << 16)

#define SIP_OPT_SDP_ANAT   (1 << 9)

#define SIP_OPT_SEC_AGREE   (1 << 10)

#define SIP_OPT_TARGET_DIALOG   (1 << 13)

#define SIP_OPT_TIMER   (1 << 2)

#define SIP_OUTGOING   (1 << 13)

#define SIP_PAGE2_ALLOWOVERLAP   (1 << 17)

#define SIP_PAGE2_ALLOWSUBSCRIBE   (1 << 16)

#define SIP_PAGE2_BUGGY_MWI   (1 << 26)

#define SIP_PAGE2_CALL_ONHOLD   (3 << 23)

#define SIP_PAGE2_CALL_ONHOLD_INACTIVE   (1 << 24)

#define SIP_PAGE2_CALL_ONHOLD_ONEDIR   (1 << 23)

#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)

#define SIP_PAGE2_FLAGS_TO_COPY

#define SIP_PAGE2_IGNOREREGEXPIRE   (1 << 10)

#define SIP_PAGE2_INC_RINGING   (1 << 19)

#define SIP_PAGE2_RFC2833_COMPENSATE   (1 << 25)

#define SIP_PAGE2_RT_FROMCONTACT   (1 << 4)

#define SIP_PAGE2_RTAUTOCLEAR   (1 << 2)

#define SIP_PAGE2_RTCACHEFRIENDS   (1 << 0)

#define SIP_PAGE2_RTSAVE_SYSNAME   (1 << 5)

#define SIP_PAGE2_RTUPDATE   (1 << 1)

#define SIP_PAGE2_SELFDESTRUCT   (1 << 14)

#define SIP_PAGE2_SUBSCRIBEMWIONLY   (1 << 18)

#define SIP_PAGE2_T38SUPPORT   (7 << 20)

#define SIP_PAGE2_T38SUPPORT_RTP   (2 << 20)

#define SIP_PAGE2_T38SUPPORT_TCP   (4 << 20)

#define SIP_PAGE2_T38SUPPORT_UDPTL   (1 << 20)

#define SIP_PAGE2_VIDEOSUPPORT   (1 << 15)

#define SIP_PENDINGBYE   (1 << 6)

#define SIP_PKT_DEBUG   (1 << 0)

#define SIP_PKT_IGNORE   (1 << 2)

#define SIP_PKT_WITH_TOTAG   (1 << 1)

#define SIP_PROG_INBAND   (3 << 25)

#define SIP_PROG_INBAND_NEVER   (0 << 25)

#define SIP_PROG_INBAND_NO   (1 << 25)

#define SIP_PROG_INBAND_YES   (2 << 25)

#define SIP_PROGRESS_SENT   (1 << 4)

#define SIP_PROMISCREDIR   (1 << 8)

#define SIP_REALTIME   (1 << 11)

#define SIP_REINVITE   (7 << 20)

#define SIP_REINVITE_UPDATE   (4 << 20)

#define SIP_RINGING   (1 << 3)

#define SIP_SENDRPID   (1 << 29)

#define SIP_TIMER_T1   500

#define SIP_TRANS_TIMEOUT   32000

#define SIP_TRUSTRPID   (1 << 9)

#define SIP_USECLIENTCODE   (1 << 12)

#define SIP_USEREQPHONE   (1 << 10)

#define sipdebug   ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG)

#define sipdebug_config   ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONFIG)

#define sipdebug_console   ast_test_flag(&global_flags[1], SIP_PAGE2_DEBUG_CONSOLE)

#define STANDARD_SIP_PORT   5060

Standard SIP port from RFC 3261. DO NOT CHANGE THIS.

#define SUPPORTED   1

#define SUPPORTED_EXTENSIONS   "replaces"

SIP Extensions we support.

#define T38FAX_FILL_BIT_REMOVAL   (1 << 0)

#define T38FAX_RATE_12000   (1 << 12)

#define T38FAX_RATE_14400   (1 << 13)

#define T38FAX_RATE_2400   (1 << 8)

#define T38FAX_RATE_4800   (1 << 9)

#define T38FAX_RATE_7200   (1 << 10)

#define T38FAX_RATE_9600   (1 << 11)

#define T38FAX_RATE_MANAGEMENT_LOCAL_TCF   (1 << 3)

#define T38FAX_RATE_MANAGEMENT_TRANSFERED_TCF   (0 << 3)

#define T38FAX_TRANSCODING_JBIG   (1 << 2)

#define T38FAX_TRANSCODING_MMR   (1 << 1)

#define T38FAX_UDP_EC_FEC   (1 << 4)

#define T38FAX_UDP_EC_NONE   (0 << 4)

#define T38FAX_UDP_EC_REDUNDANCY   (2 << 4)

#define T38FAX_VERSION   (3 << 6)

#define T38FAX_VERSION_0   (0 << 6)

#define T38FAX_VERSION_1   (1 << 6)

#define TRUE   1

#define UNLINK(element, head, prev)

#define VIDEO_CODEC_MASK   0x1fc0000

Enumerations

enum can_create_dialog { CAN_NOT_CREATE_DIALOG, CAN_CREATE_DIALOG, CAN_CREATE_DIALOG_UNSUPPORTED_METHOD }

enum check_auth_result {
  AUTH_DONT_KNOW = -100, 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
}

Authentication result from check_auth* functions. More...

enum domain_mode { SIP_DOMAIN_AUTO, SIP_DOMAIN_CONFIG }

Modes for SIP domain handling in the PBX. More...

enum invitestates {
  INV_NONE = 0, INV_CALLING = 1, INV_PROCEEDING = 2, INV_EARLY_MEDIA = 3,
  INV_COMPLETED = 4, INV_CONFIRMED = 5, INV_TERMINATED = 6, INV_CANCELLED = 7
}

States for the INVITE transaction, not the dialog. More...

enum parse_register_result { PARSE_REGISTER_FAILED, PARSE_REGISTER_UPDATE, PARSE_REGISTER_QUERY }

enum referstatus {
  REFER_IDLE, REFER_SENT, REFER_RECEIVED, REFER_CONFIRMED,
  REFER_ACCEPTED, REFER_RINGING, REFER_200OK, REFER_FAILED,
  REFER_NOAUTH
}

Parameters to know status of transfer. More...

enum sip_auth_type { PROXY_AUTH = 407, WWW_AUTH = 401 }

Authentication types - proxy or www authentication. More...

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

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_PING
}

SIP Request methods known by Asterisk. More...

enum sipregistrystate {
  REG_STATE_UNREGISTERED = 0, REG_STATE_REGSENT, REG_STATE_AUTHSENT, REG_STATE_REGISTERED,
  REG_STATE_REJECTED, REG_STATE_TIMEOUT, REG_STATE_NOAUTH, REG_STATE_FAILED
}

States for outbound registrations (with register= lines in sip.conf. More...

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

enum t38state {
  T38_DISABLED = 0, T38_LOCAL_DIRECT, T38_LOCAL_REINVITE, T38_PEER_DIRECT,
  T38_PEER_REINVITE, T38_ENABLED
}

T38 States for a call. More...

enum transfermodes { TRANSFER_OPENFORALL, TRANSFER_CLOSED }

Authorization scheme for call transfers. More...

enum xmittype { XMIT_CRITICAL = 2, XMIT_RELIABLE = 1, XMIT_UNRELIABLE = 0 }

Functions

static const char * __get_header (const struct sip_request *req, const char *name, int *start)

static void __sip_ack (struct sip_pvt *p, int seqno, int resp, int sipmethod)

Acknowledges receipt of a packet and stops retransmission.

static int __sip_autodestruct (void *data)

Kill a SIP dialog (called by scheduler).

static void __sip_destroy (struct sip_pvt *p, int lockowner, int lockdialoglist)

Execute destruction of SIP dialog structure, release memory.

static int __sip_do_register (struct sip_registry *r)

Register with SIP proxy.

static void __sip_pretend_ack (struct sip_pvt *p)

Pretend to ack all packets maybe the lock on p is not strictly necessary but there might be a race.

static int __sip_reliable_xmit (struct sip_pvt *p, int seqno, int resp, char *data, int len, int fatal, int sipmethod)

Transmit packet with retransmits.

static int __sip_semi_ack (struct sip_pvt *p, int seqno, int resp, int sipmethod)

Acks receipt of packet, keep it around (used for provisional responses).

static int __sip_show_channels (int fd, int argc, char *argv[], int subscriptions)

SIP show channels CLI (main function).

static int __sip_xmit (struct sip_pvt *p, char *data, int len)

Transmit SIP message.

static int __transmit_response (struct sip_pvt *p, const char *msg, const struct sip_request *req, enum xmittype reliable)

Base transmit response function.

static int _sip_show_peer (int type, int fd, struct mansession *s, const struct message *m, int argc, const char *argv[])

Show one peer in detail (main function).

static int _sip_show_peers (int fd, int *total, struct mansession *s, const struct message *m, int argc, const char *argv[])

_sip_show_peers: Execute sip show peers command

static void add_blank (struct sip_request *req)

add a blank line if no body

static void add_codec_to_sdp (const struct sip_pvt *p, int codec, int sample_rate, char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, int debug, int *min_packet_size)

Add codec offer to SDP offer/answer body in INVITE or 200 OK.

static int add_digit (struct sip_request *req, char digit, unsigned int duration)

Add DTMF INFO tone to sip message.

static int add_header (struct sip_request *req, const char *var, const char *value)

Add header to SIP message.

static int add_header_contentLength (struct sip_request *req, int len)

Add 'Content-Length' header to SIP message.

static int add_line (struct sip_request *req, const char *line)

Add content (not header) to SIP message.

static void add_noncodec_to_sdp (const struct sip_pvt *p, int format, int sample_rate, char **m_buf, size_t *m_size, char **a_buf, size_t *a_size, int debug)

Add RFC 2833 DTMF offer to SDP.

static struct sip_auth * add_realm_authentication (struct sip_auth *authlist, char *configuration, int lineno)

Add realm authentication in list.

static void add_route (struct sip_request *req, struct sip_route *route)

Add route header into request per learned route.

static enum sip_result add_sdp (struct sip_request *resp, struct sip_pvt *p)

Add Session Description Protocol message.

static int add_sip_domain (const char *domain, const enum domain_mode mode, const char *context)

Add SIP domain to list of domains we are responsible for.

static int add_t38_sdp (struct sip_request *resp, struct sip_pvt *p)

Add T.38 Session Description Protocol message.

static int add_text (struct sip_request *req, const char *text)

Add text body to SIP message.

static struct ast_variable * add_var (const char *buf, struct ast_variable *list)

static int add_vidupdate (struct sip_request *req)

add XML encoded media control with update

static void append_date (struct sip_request *req)

Append date to SIP message.

static void append_history_full (struct sip_pvt *p, const char *fmt,...)

Append to SIP dialog history with arg list.

static void static void append_history_va (struct sip_pvt *p, const char *fmt, va_list ap)

Append to SIP dialog history with arg list.

AST_LIST_HEAD_NOLOCK (sip_history_head, sip_history)

static AST_LIST_HEAD_STATIC (domain_list, domain)

AST_MODULE_INFO (ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT,"Session Initiation Protocol (SIP)",.load=load_module,.unload=unload_module,.reload=reload,)

AST_MUTEX_DEFINE_STATIC (dialoglock)

Protect the SIP dialog list (of sip_pvt's).

AST_MUTEX_DEFINE_STATIC (sip_reload_lock)

AST_MUTEX_DEFINE_STATIC (monlock)

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)

static void ast_quiet_chan (struct ast_channel *chan)

Turn off generator data XXX Does this function belong in the SIP channel?

static int ast_sip_ouraddrfor (struct in_addr *them, struct in_addr *us)

NAT fix - decide which IP address to use for ASterisk server?

AST_THREADSTORAGE_CUSTOM (ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup)

A per-thread temporary pvt structure.

static int attempt_transfer (struct sip_dual *transferer, struct sip_dual *target)

Attempt transfer of SIP call This fix for attended transfers on a local PBX.

static void auth_headers (enum sip_auth_type code, char **header, char **respheader)

return the request and response heade for a 401 or 407 code

static int auto_congest (void *nothing)

Scheduled congestion on a call.

static void build_callid_pvt (struct sip_pvt *pvt)

Build SIP Call-ID value for a non-REGISTER transaction.

static void build_callid_registry (struct sip_registry *reg, struct in_addr ourip, const char *fromdomain)

Build SIP Call-ID value for a REGISTER transaction.

static void build_contact (struct sip_pvt *p)

Build contact header - the contact header we send out.

static struct sip_peer * build_peer (const char *name, struct ast_variable *v, struct ast_variable *alt, int realtime)

Build peer from configuration (file or realtime static/dynamic).

static int build_reply_digest (struct sip_pvt *p, int method, char *digest, int digest_len)

Build reply digest.

static void build_route (struct sip_pvt *p, struct sip_request *req, int backwards)

Build route list from Record-Route header.

static void build_rpid (struct sip_pvt *p)

Build the Remote Party-ID & From using callingpres options.

static struct sip_user * build_user (const char *name, struct ast_variable *v, int realtime)

Initiate a SIP user structure from configuration (configuration or realtime).

static void build_via (struct sip_pvt *p)

Build a Via header for a request.

static int cb_extensionstate (char *context, char *exten, int state, void *data)

Callback for the devicestate notification (SUBSCRIBE) support subsystem.

static enum check_auth_result check_auth (struct sip_pvt *p, struct sip_request *req, const char *username, const char *secret, const char *md5secret, int sipmethod, char *uri, enum xmittype reliable, int ignore)

Check user authorization from peer definition Some actions, like REGISTER and INVITEs from peers require authentication (if peer have secret set).

static enum check_auth_result check_peer_ok (struct sip_pvt *p, char *of, struct sip_request *req, int sipmethod, struct sockaddr_in *sin, struct sip_peer **authpeer, enum xmittype reliable, char *rpid_num, char *calleridname, char *uri2)

Validate peer authentication.

static void check_pendings (struct sip_pvt *p)

Check pending actions on SIP call.

static void check_rtp_timeout (struct sip_pvt *dialog, time_t t)

helper function for the monitoring thread

static int check_sip_domain (const char *domain, char *context, size_t len)

check_sip_domain: Check if domain part of uri is local to our server

static int check_user (struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin)

Find user If we get a match, this will add a reference pointer to the user object in ASTOBJ, that needs to be unreferenced.

static enum check_auth_result check_user_full (struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, enum xmittype reliable, struct sockaddr_in *sin, struct sip_peer **authpeer)

Check if matching user or peer is defined Match user on From: user name and peer on IP/port This is used on first invite (not re-invites) and subscribe requests.

static enum check_auth_result check_user_ok (struct sip_pvt *p, char *of, struct sip_request *req, int sipmethod, struct sockaddr_in *sin, enum xmittype reliable, char *rpid_num, char *calleridname, char *uri2)

Validate user authentication.

static void check_via (struct sip_pvt *p, struct sip_request *req)

check Via: header for hostname, port and rport request/answer

static void cleanup_stale_contexts (char *new, char *old)

Destroy disused contexts between reloads Only used in reload_config so the code for regcontext doesn't get ugly.

static int clear_realm_authentication (struct sip_auth *authlist)

Clear realm authentication list (at reload).

static void clear_sip_domains (void)

Clear our domain list (at reload).

static char * complete_sip_debug_peer (const char *line, const char *word, int pos, int state)

Support routine for 'sip debug peer' CLI.

static char * complete_sip_peer (const char *word, int state, int flags2)

Do completion on peer name.

static char * complete_sip_prune_realtime_peer (const char *line, const char *word, int pos, int state)

Support routine for 'sip prune realtime peer' CLI.

static char * complete_sip_prune_realtime_user (const char *line, const char *word, int pos, int state)

Support routine for 'sip prune realtime user' CLI.

static char * complete_sip_show_peer (const char *line, const char *word, int pos, int state)

Support routine for 'sip show peer' CLI.

static char * complete_sip_show_user (const char *line, const char *word, int pos, int state)

Support routine for 'sip show user' CLI.

static char * complete_sip_user (const char *word, int state, int flags2)

Do completion on user name.

static char * complete_sipch (const char *line, const char *word, int pos, int state)

Support routine for 'sip show channel' CLI.

static char * complete_sipnotify (const char *line, const char *word, int pos, int state)

Support routine for 'sip notify' CLI.

static int copy_all_header (struct sip_request *req, const struct sip_request *orig, const char *field)

Copy all headers from one request to another.

static int copy_header (struct sip_request *req, const struct sip_request *orig, const char *field)

Copy one header field from one request to another.

static void copy_request (struct sip_request *dst, const struct sip_request *src)

copy SIP request (mostly used to save request for responses)

static struct ast_variable * copy_vars (struct ast_variable *src)

static int copy_via_headers (struct sip_pvt *p, struct sip_request *req, const struct sip_request *orig, const char *field)

Copy SIP VIA Headers from the request to the response.

static int create_addr (struct sip_pvt *dialog, const char *opeer)

create address structure from peer name Or, if peer not found, find it in the global DNS returns TRUE (-1) on failure, FALSE on success

static int create_addr_from_peer (struct sip_pvt *dialog, struct sip_peer *peer)

Create address structure from peer reference. return -1 on error, 0 on success.

static void destroy_association (struct sip_peer *peer)

Remove registration data from realtime database or AST/DB when registration expires.

Asterisk developer's documentation :: Codename Pineapple

chan_sip.c File Reference

Detailed Description

Data Structures

Defines

Enumerations

Functions