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More doxygen comments.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128485 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2008-07-06 18:25:39 +00:00
parent fce9af0fa0
commit 40c0e6ed47
1 changed files with 52 additions and 37 deletions
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89
channels/chan_sip.c
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@@ -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 thie moment this is set together with sip_debug_console.
* It should either go away or be implemented properly.
* At the 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