Adding doxygen comments to missing parts, moving some #define
...trying to get my head around the thoughts behind the TCP/TLS stuff and figure out what needs to be done to make it useful... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@ -1372,6 +1372,10 @@ struct sip_pvt {
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*/
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struct ao2_container *dialogs;
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#define sip_pvt_lock(x) ao2_lock(x)
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#define sip_pvt_trylock(x) ao2_trylock(x)
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#define sip_pvt_unlock(x) ao2_unlock(x)
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/*!
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* when we create or delete references, make sure to use these
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* functions so we keep track of the refcounts.
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@ -1768,8 +1772,8 @@ static struct sockaddr_in stunaddr; /*!< stun server address */
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*/
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static struct ast_ha *localaddr; /*!< List of local networks, on the same side of NAT as this Asterisk */
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static int ourport_tcp;
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static int ourport_tls;
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static int ourport_tcp; /*!< The port used for TCP connections */
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static int ourport_tls; /*!< The port used for TCP/TLS connections */
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static struct sockaddr_in debugaddr;
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static struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */
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@ -2069,6 +2073,10 @@ static int get_destination(struct sip_pvt *p, struct sip_request *oreq);
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static int get_msg_text(char *buf, int len, struct sip_request *req, int addnewline);
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static int transmit_state_notify(struct sip_pvt *p, int state, int full, int timeout);
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/*-- TCP connection handling ---*/
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static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct ast_tcptls_session_instance *ser);
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static void *sip_tcp_worker_fn(void *);
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/*--- Constructing requests and responses */
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static void initialize_initreq(struct sip_pvt *p, struct sip_request *req);
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static int init_req(struct sip_request *req, int sipmethod, const char *recip);
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@ -2188,11 +2196,14 @@ static const struct ast_channel_tech sip_tech = {
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*/
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static struct ast_channel_tech sip_tech_info;
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static void *sip_tcp_worker_fn(void *);
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/*! \brief Working TLS connection configuration */
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static struct ast_tls_config sip_tls_cfg;
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/*! \brief Default TLS connection configuration */
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static struct ast_tls_config default_tls_cfg;
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/*! \brief The TCP server definition */
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static struct server_args sip_tcp_desc = {
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.accept_fd = -1,
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.master = AST_PTHREADT_NULL,
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@ -2203,6 +2214,7 @@ static struct server_args sip_tcp_desc = {
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.worker_fn = sip_tcp_worker_fn,
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};
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/*! \brief The TCP/TLS server definition */
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static struct server_args sip_tls_desc = {
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.accept_fd = -1,
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.master = AST_PTHREADT_NULL,
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@ -2253,8 +2265,8 @@ static struct ast_rtp_protocol sip_rtp = {
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.get_codec = sip_get_codec,
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};
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static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct ast_tcptls_session_instance *ser);
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/*! \brief SIP TCP connection handler */
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static void *sip_tcp_worker_fn(void *data)
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{
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struct ast_tcptls_session_instance *ser = data;
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@ -2262,7 +2274,7 @@ static void *sip_tcp_worker_fn(void *data)
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return _sip_tcp_helper_thread(NULL, ser);
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}
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/*! \brief SIP TCP helper function */
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/*! \brief SIP TCP thread management function */
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static void *_sip_tcp_helper_thread(struct sip_pvt *pvt, struct ast_tcptls_session_instance *ser)
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{
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int res, cl;
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@ -2355,8 +2367,10 @@ cleanup2:
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fclose(ser->f);
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ser->f = NULL;
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ser->fd = -1;
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if (reqcpy.data)
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if (reqcpy.data) {
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ast_free(reqcpy.data);
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}
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if (req.data) {
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ast_free(req.data);
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req.data = NULL;
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@ -2369,9 +2383,6 @@ cleanup2:
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return NULL;
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}
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#define sip_pvt_lock(x) ao2_lock(x)
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#define sip_pvt_trylock(x) ao2_trylock(x)
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#define sip_pvt_unlock(x) ao2_unlock(x)
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/*!
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* helper functions to unreference various types of objects.
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@ -4286,6 +4297,9 @@ static int create_addr(struct sip_pvt *dialog, const char *opeer, struct sockadd
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/* Let's see if we can find the host in DNS. First try DNS SRV records,
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then hostname lookup */
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/*! \todo Fix this function. When we ask SRC, we should check all transports
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In the future, we should first check NAPTR to find out transport preference
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*/
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hostn = peername;
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portno = port ? atoi(port) : (dialog->socket.type & SIP_TRANSPORT_TLS) ? STANDARD_TLS_PORT : STANDARD_SIP_PORT;
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if (global_srvlookup) {
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@ -10281,7 +10295,10 @@ static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, st
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ast_sched_add(sched, (expiry + 10) * 1000, expire_register, peer);
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pvt->expiry = expiry;
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snprintf(data, sizeof(data), "%s:%d:%d:%s:%s", ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port), expiry, peer->username, peer->fullcontact);
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/* Saving TCP connections is useless, we won't be able to reconnect */
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/* Saving TCP connections is useless, we won't be able to reconnect
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XXX WHY???? XXX
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\todo check this
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*/
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if (!peer->rt_fromcontact && (peer->socket.type & SIP_TRANSPORT_UDP))
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ast_db_put("SIP/Registry", peer->name, data);
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manager_event(EVENT_FLAG_SYSTEM, "PeerStatus", "ChannelType: SIP\r\nPeer: SIP/%s\r\nPeerStatus: Registered\r\nAddress: %s\r\nPort: %d\r\n", peer->name, ast_inet_ntoa(peer->addr.sin_addr), ntohs(peer->addr.sin_port));
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