@ -293,9 +293,10 @@ enum transfermodes {
} ;
/*! \brief The result of a lot of functions */
enum sip_result {
AST_SUCCESS = 0 ,
AST_FAILURE = - 1 ,
AST_SUCCESS = 0 , /*! FALSE means success, funny enough */
AST_FAILURE = - 1 ,
} ;
/*! \brief States for the INVITE transaction, not the dialog
@ -329,6 +330,9 @@ static const struct invstate2stringtable {
{ INV_CANCELLED , " Cancelled " }
} ;
/*! \brief When sending a SIP message, we can send with a few options, depending on
type of SIP request . UNRELIABLE is moslty used for responses to repeated requests ,
where the original response would be sent RELIABLE in an INVITE transaction */
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 */
@ -342,6 +346,7 @@ enum parse_register_result {
PARSE_REGISTER_QUERY ,
} ;
/*! \brief Type of subscription, based on the packages we do support */
enum subscriptiontype {
NONE = 0 ,
XPIDF_XML ,
@ -449,11 +454,13 @@ enum st_refresher {
SESSION_TIMER_REFRESHER_UAS /*!< Session is refreshed by the UAS */
} ;
/*!< Define some SIP transports */
/*! \brief Define some implemented SIP transports
\ note Asterisk does not support SCTP or UDP / DTLS
*/
enum sip_transport {
SIP_TRANSPORT_UDP = 1 ,
SIP_TRANSPORT_TCP = 1 < < 1 ,
SIP_TRANSPORT_TLS = 1 < < 2 ,
SIP_TRANSPORT_UDP = 1 , /*!< Unreliable transport for SIP, needs retransmissions */
SIP_TRANSPORT_TCP = 1 < < 1 , /*!< Reliable, but unsecure */
SIP_TRANSPORT_TLS = 1 < < 2 , /*!< TCP/TLS - reliable and secure transport for signalling */
} ;
/*! \brief definition of a sip proxy server
@ -640,12 +647,18 @@ static const struct cfsip_options {
*/
# define ALLOWED_METHODS "INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY"
/*! \brief SIP Extensions we support */
/*! \brief SIP Extensions we support
\ note This should be generated based on the previous array
in combination with settings .
\ todo We should not have " timer " if it ' s disabled in the configuration file .
*/
# define SUPPORTED_EXTENSIONS "replaces, timer"
/*! \brief Standard SIP and TLS port from RFC 3261. DO NOT CHANGE THIS */
/*! \brief Standard SIP unsecure port for UDP and TCP from RFC 3261. DO NOT CHANGE THIS */
# define STANDARD_SIP_PORT 5060
/*! \brief Standard SIP TLS port for sips: from RFC 3261. DO NOT CHANGE THIS */
# define STANDARD_TLS_PORT 5061
/*! \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 ,
@ -676,12 +689,12 @@ static const struct cfsip_options {
# 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_TOS_TEXT 0 /*!< Text packets should be marked as XXXX XXXX, but the default is 0 to be compatible with previous versions. */
# define DEFAULT_COS_SIP 4
# define DEFAULT_COS_AUDIO 5
# define DEFAULT_COS_VIDEO 6
# define DEFAULT_COS_TEXT 5
# define DEFAULT_ALLOW_EXT_DOM TRUE
# define DEFAULT_REALM "asterisk"
# define DEFAULT_COS_SIP 4 /*!< Level 2 class of service for SIP signalling */
# define DEFAULT_COS_AUDIO 5 /*!< Level 2 class of service for audio media */
# define DEFAULT_COS_VIDEO 6 /*!< Level 2 class of service for video media */
# define DEFAULT_COS_TEXT 5 /*!< Level 2 class of service for text media (T.140) */
# define DEFAULT_ALLOW_EXT_DOM TRUE /*!< Allow external domains */
# define DEFAULT_REALM "asterisk" /*!< Realm for HTTP digest authentication */
# define DEFAULT_NOTIFYRINGING TRUE
# define DEFAULT_PEDANTIC FALSE
# define DEFAULT_AUTOCREATEPEER FALSE
@ -692,7 +705,7 @@ static const struct cfsip_options {
# ifndef DEFAULT_USERAGENT
# define DEFAULT_USERAGENT "Asterisk PBX" /*!< Default Useragent: header unless re-defined in sip.conf */
# define DEFAULT_SDPSESSION "Asterisk PBX" /*!< Default SDP session name, (s=) header unless re-defined in sip.conf */
# define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
# define DEFAULT_SDPOWNER "root" /*!< Default SDP username field in (o=) header unless re-defined in sip.conf */
# endif
/*@}*/
@ -773,17 +786,18 @@ static int global_callevents; /*!< Whether we send manager events or not */
static int global_authfailureevents ; /*!< Whether we send authentication failure manager events or not. Default no. */
static int global_t1 ; /*!< T1 time */
static int global_t1min ; /*!< T1 roundtrip time minimum */
static int global_timer_b ; /*!< Timer B - RFC 3261 Section 17.1.1.2 */
static int global_regextenonqualify ; /*!< Whether to add/remove regexten when qualifying peers */
static int global_autoframing ; /*!< Turn autoframing on or off. */
static int global_timer_b ; /*!< Timer B - RFC 3261 Section 17.1.1.2 */
static int global_regextenonqualify ; /*!< Whether to add/remove regexten when qualifying peers */
static int global_autoframing ; /*!< Turn autoframing on or off. */
static enum transfermodes global_allowtransfer ; /*!< SIP Refer restriction scheme */
static struct sip_proxy global_outboundproxy ; /*!< Outbound proxy */
static int global_matchexterniplocally ; /*!< Match externip/externhost setting against localnet setting */
static int global_qualifyfreq ; /*!< Qualify frequency */
static int global_matchexterniplocally ; /*!< Match externip/externhost setting against localnet setting */
static int global_qualifyfreq ; /*!< Qualify frequency */
/*! \brief Codecs that we support by default: */
static int global_capability = AST_FORMAT_ULAW | AST_FORMAT_ALAW | AST_FORMAT_GSM | AST_FORMAT_H263 ;
static enum st_mode global_st_mode ; /*!< Mode of operation for Session-Timers */
static enum st_refresher global_st_refresher ; /*!< Session-Timer refresher */
static int global_min_se ; /*!< Lowest threshold for session refresh interval */
@ -801,14 +815,13 @@ static int regobjs = 0; /*!< Registry objects */
/* }@ */
static struct ast_flags global_flags [ 2 ] = { { 0 } } ; /*!< global SIP_ flags */
static char used_context [ AST_MAX_CONTEXT ] ; /*!< name of automatically created context for unloading */
static char used_context [ AST_MAX_CONTEXT ] ; /*!< name of automatically created context for unloading */
AST_MUTEX_DEFINE_STATIC ( netlock ) ;
/*! \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 ( monlock ) ;
AST_MUTEX_DEFINE_STATIC ( sip_reload_lock ) ;
@ -829,12 +842,12 @@ static int *sipsock_read_id; /*!< ID of IO entry for sipsock FD */
# define DEC_CALL_RINGING 2
# define INC_CALL_RINGING 3
/*! < The SIP socket definition */
/*! \brief The SIP socket definition */
struct sip_socket {
enum sip_transport type ;
int fd ;
enum sip_transport type ; /*!< UDP, TCP or TLS */
int fd ; /*!< Filed descriptor, the actual socket */
uint16_t port ;
struct ast_tcptls_session_instance * ser ;
struct ast_tcptls_session_instance * ser ; /* If tcp or tls, a socket manager */
} ;
/*! \brief sip_request: The data grabbed from the UDP socket
@ -1097,8 +1110,8 @@ enum sip_debug_e {
static enum sip_debug_e sipdebug ;
/*! \brief extra debugging for 'text' related events.
* At th i e moment this is set together with sip_debug_console .
* It should either go away or be implemented properly .
* At th e moment this is set together with sip_debug_console .
* \ note It should either go away or be implemented properly .
*/
static int sipdebug_text ;
@ -1210,7 +1223,7 @@ struct sip_st_cfg {
/*! \brief sip_pvt: structures used for each SIP dialog, ie. a call, a registration, a subscribe.
/*! \brief Structure used for each SIP dialog, ie. a call, a registration, a subscribe.
* Created and initialized by sip_alloc ( ) , the descriptor goes into the list of
* descriptors ( dialoglist ) .
*/
@ -1364,7 +1377,7 @@ struct sip_pvt {
/*! Max entires in the history list for a sip_pvt */
# define MAX_HISTORY_ENTRIES 50
/*!
/*! \brief
* Here we implement the container for dialogs ( sip_pvt ) , defining
* generic wrapper functions to ease the transition from the current
* implementation ( a single linked list ) to a different container .
@ -1378,7 +1391,7 @@ struct ao2_container *dialogs;
# define sip_pvt_trylock(x) ao2_trylock(x)
# define sip_pvt_unlock(x) ao2_unlock(x)
/*!
/*! \brief
* when we create or delete references , make sure to use these
* functions so we keep track of the refcounts .
* To simplify the code , we allow a NULL to be passed to dialog_unref ( ) .
@ -1386,6 +1399,7 @@ struct ao2_container *dialogs;
# ifdef REF_DEBUG
# define dialog_ref(arg1,arg2) dialog_ref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
# define dialog_unref(arg1,arg2) dialog_unref_debug((arg1),(arg2), __FILE__, __LINE__, __PRETTY_FUNCTION__)
static struct sip_pvt * dialog_ref_debug ( struct sip_pvt * p , char * tag , char * file , int line , const char * func )
{
if ( p )
@ -1567,7 +1581,7 @@ struct sip_registry {
AST_STRING_FIELD ( callback ) ; /*!< Contact extension */
AST_STRING_FIELD ( random ) ;
) ;
enum sip_transport transport ;
enum sip_transport transport ; /*!< Transport for this registration UDP, TCP or TLS */
int portno ; /*!< Optional port override */
int expire ; /*!< Sched ID of expiration */
int expiry ; /*!< Value to use for the Expires header */
@ -1585,11 +1599,12 @@ struct sip_registry {
char lastmsg [ 256 ] ; /*!< Last Message sent/received */
} ;
/*! \brief Definition of a thread that handles a socket */
struct sip_threadinfo {
int stop ;
pthread_t threadid ;
struct ast_tcptls_session_instance * ser ;
enum sip_transport type ; /* We keep a copy of the type here so we can display it in the connection list */
enum sip_transport type ; /* !< We keep a copy of the type here so we can display it in the connection list */
AST_LIST_ENTRY ( sip_threadinfo ) list ;
} ;
@ -1600,7 +1615,7 @@ static int hash_peer_size = 17;
static int hash_dialog_size = 17 ;
static int hash_user_size = 17 ;
# else
static int hash_peer_size = 563 ;
static int hash_peer_size = 563 ; /*!< Size of peer hash table, prime number preferred! */
static int hash_dialog_size = 563 ;
static int hash_user_size = 563 ;
# endif
@ -1608,7 +1623,7 @@ static int hash_user_size = 563;
/*! \brief The thread list of TCP threads */
static AST_LIST_HEAD_STATIC ( threadl , sip_threadinfo ) ;
/*! \brief The peer list: Peers and Friends */
/*! \brief The peer list: Users, Peers and Friends */
struct ao2_container * peers ;
struct ao2_container * peers_by_ip ;
@ -1618,7 +1633,7 @@ static struct ast_register_list {
int recheck ;
} regl ;
/*!
/*! \brief
* \ note The only member of the peer used here is the name field
*/
static int peer_hash_cb ( const void * obj , const int flags )
@ -1731,7 +1746,7 @@ static struct sip_auth *authl = NULL;
*/
static int sipsock = - 1 ;
static struct sockaddr_in bindaddr ; /*!< The address we bind to */
static struct sockaddr_in bindaddr ; /*!< UDP: The address we bind to */
/*! \brief our (internal) default address/port to put in SIP/SDP messages
* internip is initialized picking a suitable address from one of the
Olle Johansson
17 years ago