From 41643411561f8e86d058be248d99dd273bdcaf74 Mon Sep 17 00:00:00 2001
From: Yun Liu <yun@yun.(none)>
Date: Tue, 23 Sep 2008 10:20:30 -0400
Subject: [PATCH] sip manager implementation including all the callbacks
---
src/sipmanager.cpp | 1175 ++++++++++++++++++++++++++++++++++++++++++++
src/sipmanager.h | 152 ++++++
2 files changed, 1327 insertions(+)
create mode 100755 src/sipmanager.cpp
create mode 100755 src/sipmanager.h
diff --git a/src/sipmanager.cpp b/src/sipmanager.cpp
new file mode 100755
index 0000000000..854d0ee0e9
--- /dev/null
+++ b/src/sipmanager.cpp
@@ -0,0 +1,1175 @@
+/*
+ * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
+ * Author: Yun Liu <yun.liu@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+
+#include "manager.h"
+#include "sipcall.h"
+#include "sipmanager.h"
+#include "sipvoiplink.h"
+
+#define DEFAULT_SIP_PORT 5068
+#define RANDOM_SIP_PORT rand() % 64000 + 1024
+#define RANDOM_LOCAL_PORT ((rand() % 27250) + 5250)*2
+
+SIPManager *SIPManager::_current;
+pjsip_inv_session* SIPManager::_invSession;
+
+SIPManager::SIPManager() {
+ _useStun = false;
+ _localIPAddress = "127.0.0.1";
+ SIPManager::_current = this;
+}
+
+SIPManager::~SIPManager() {
+ _debug("Somebody call me!\n");
+}
+
+pj_status_t SIPManager::sipCreate() {
+ pj_status_t status;
+ //_debug("stun host is %s\n", _stunHost.ptr);
+
+ /* Init pjsua data */
+ //init_data();
+
+ /* Set default logging settings */
+ //pjsua_logging_config_default(&pjsua_var.log_cfg);
+
+ /* Init PJLIB: */
+ status = pj_init();
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Could not initialize PJSip\n");
+ return status;
+ }
+
+ /* Init random seed */
+ //init_random_seed();
+
+ /* Init PJLIB-UTIL: */
+ status = pjlib_util_init();
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Could not initialize PJ Util\n");
+ return status;
+ }
+
+ /* Init PJNATH */
+ status = pjnath_init();
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Could not initialize PJ Nath\n");
+ return status;
+ }
+
+ /* Set default sound device ID */
+ //pjsua_var.cap_dev = pjsua_var.play_dev = -1;
+
+ /* Init caching pool. */
+ pj_caching_pool_init(&_cp, NULL, 0);
+
+ /* Create memory pool for application. */
+ _pool = pj_pool_create(&_cp.factory, "sflphone", 4000, 4000, NULL);
+
+ if (!_pool) {
+ _debug("! SIP Failure: Could not initialize memory pool\n");
+ return PJ_ENOMEM;
+ }
+
+ /* Create mutex */
+ status = pj_mutex_create_recursive(_pool, "sflphone", &_mutex);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to create mutex\n");
+ return status;
+ }
+
+ /* Must create SIP endpoint to initialize SIP parser. The parser
+ * is needed for example when application needs to call pjsua_verify_url().
+ */
+ status = pjsip_endpt_create(&_cp.factory,
+ pj_gethostname()->ptr,
+ &_endpt);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to create mutex\n");
+ return status;
+ }
+
+ return PJ_SUCCESS;
+}
+
+pj_status_t SIPManager::sipInit() {
+ pj_status_t status;
+ const pj_str_t STR_OPTIONS = {"OPTIONS", 7};
+
+ /* Init SIP UA: */
+
+ //FIXME! DNS initialize here! */
+
+ /* Start resolving STUN server */
+ // if we useStun and we failed to receive something on port 5060, we try a random port
+ // If use STUN server, firewall address setup
+ if (!loadSIPLocalIP()) {
+ _debug("! SIP FAILURE: Unable to determine network capabilities\n");
+ return false;
+ }
+ int errPjsip = 0;
+ int port = DEFAULT_SIP_PORT;
+
+ //_debug("stun host is %s\n", _stunHost.ptr);
+ if (_useStun && !Manager::instance().behindNat(_stunServer, port)) {
+ port = RANDOM_SIP_PORT;
+ if (!Manager::instance().behindNat(_stunServer, port)) {
+ _debug("! SIP Failure: Unable to check NAT setting\n");
+ return false; // hoho we can't use the random sip port too...
+ }
+ }
+
+ _localPort = port;
+ if (_useStun) {
+ // set by last behindNat() call (ish)...
+ stunServerResolve();
+ _localExternAddress = Manager::instance().getFirewallAddress();
+ _localExternPort = Manager::instance().getFirewallPort();
+ } else {
+ _localExternAddress = _localIPAddress;
+ _localExternPort = _localPort;
+ }
+
+ errPjsip = createUDPServer();
+ if (errPjsip != 0) {
+ _debug("* SIP Info: could not initialize SIP listener on port %d\n", port);
+ port = RANDOM_SIP_PORT;
+ _debug("! SIP Failure: SIP failed to listen on port %d\n", port);
+ return errPjsip;
+ }
+ _debug(" SIP Init: listening on port %d\n", _localExternPort);
+
+ /* Initialize transaction layer: */
+ status = pjsip_tsx_layer_init_module(_endpt);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to initialize transaction layer.\n");
+ return status;
+ }
+
+ /* Initialize UA layer module: */
+ status = pjsip_ua_init_module(_endpt, NULL);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to initialize UA layer module.\n");
+ return status;
+ }
+
+
+ /* Initialize Replaces support. */
+ status = pjsip_replaces_init_module(_endpt);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to initialize Replaces support.\n");
+ return status;
+ }
+
+ /* Initialize 100rel support */
+ status = pjsip_100rel_init_module(_endpt);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to initialize 100rel support.\n");
+ return status;
+ }
+
+ /* Initialize and register sflphone module. */
+ {
+ const pjsip_module mod_initializer ={
+ NULL, NULL, /* prev, next. */
+ { "mod-sflphone", 9
+ }, /* Name. */
+ -1, /* Id */
+ PJSIP_MOD_PRIORITY_APPLICATION, /* Priority */
+ NULL, /* load() */
+ NULL, /* start() */
+ NULL, /* stop() */
+ NULL, /* unload() */
+ &mod_on_rx_request, /* on_rx_request() */
+ &mod_on_rx_response, /* on_rx_response() */
+ NULL, /* on_tx_request. */
+ NULL, /* on_tx_response() */
+ NULL, /* on_tsx_state() */
+ };
+
+ _mod = mod_initializer;
+
+ status = pjsip_endpt_register_module(_endpt, &_mod);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to register sflphone module.\n");
+ return status;
+ }
+ }
+
+
+
+ /* Initialize PJSUA call subsystem: */
+ /*status = pjsua_call_subsys_init(ua_cfg);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to initialize ")
+ goto on_error;
+ }*/
+
+
+ /* Initialize PJSUA media subsystem */
+ /*status = pjsua_media_subsys_init(media_cfg);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to initialize ")
+ goto on_error;
+ }*/
+
+
+ /* Init core SIMPLE module : */
+ status = pjsip_evsub_init_module(_endpt);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to initialize core SIMPLE module.\n");
+ return status;
+ }
+
+
+ /* Init presence module: */
+ status = pjsip_pres_init_module(_endpt, pjsip_evsub_instance());
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to initialize presence module.\n");
+ return status;
+ }
+
+ /* Init PUBLISH module */
+ pjsip_publishc_init_module(_endpt);
+
+ /* Init xfer/REFER module */
+ status = pjsip_xfer_init_module(_endpt);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to initialize xfer/PEFER module.\n");
+ return status;
+ }
+
+ /* Init pjsua presence handler: */
+ /*status = pjsua_pres_init();
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to initialize ")
+ goto on_error;
+ }*/
+
+ /* Init out-of-dialog MESSAGE request handler. */
+ /*status = pjsua_im_init();
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to initialize ")
+ goto on_error;
+ }*/
+
+ {
+ const pjsip_module handler ={
+ NULL, NULL, /* prev, next. */
+ { "mod-pjsua-options", 17
+ }, /* Name. */
+ -1, /* Id */
+ PJSIP_MOD_PRIORITY_APPLICATION, /* Priority */
+ NULL, /* load() */
+ NULL, /* start() */
+ NULL, /* stop() */
+ NULL, /* unload() */
+ &options_on_rx_request, /* on_rx_request() */
+ NULL, /* on_rx_response() */
+ NULL, /* on_tx_request. */
+ NULL, /* on_tx_response() */
+ NULL, /* on_tsx_state() */
+
+ };
+ _options_handler = handler;
+ }
+
+ /* Register OPTIONS handler */
+ pjsip_endpt_register_module(_endpt, &_options_handler);
+
+ /* Add OPTIONS in Allow header */
+ pjsip_endpt_add_capability(_endpt, NULL, PJSIP_H_ALLOW,
+ NULL, 1, &STR_OPTIONS);
+
+
+ // Initialize invite session module
+ // These callbacks will be called on incoming requests, media session state, etc.
+ {
+ pjsip_inv_callback inv_cb;
+
+ // Init the callback for INVITE session:
+ pj_bzero(&inv_cb, sizeof (inv_cb));
+
+ inv_cb.on_state_changed = &call_on_state_changed;
+ inv_cb.on_new_session = &call_on_forked;
+ inv_cb.on_media_update = &call_on_media_update;
+ inv_cb.on_tsx_state_changed = &call_on_tsx_changed;
+
+ _debug("AAAA VOIP callbacks initialized\n");
+
+ // Initialize session invite module
+ status = pjsip_inv_usage_init(_endpt, &inv_cb);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ }
+
+
+ // Add endpoint capabilities (INFO, OPTIONS, etc) for this UA
+ {
+ pj_str_t allowed[] = {
+ {"INFO", 4},
+ {"REGISTER", 8
+ }
+ }; // //{"INVITE", 6}, {"ACK",3}, {"BYE",3}, {"CANCEL",6}, {"OPTIONS", 7},
+ pj_str_t accepted = {"application/sdp", 15
+ };
+
+ // Register supported methods
+ pjsip_endpt_add_capability(_endpt, &_mod, PJSIP_H_ALLOW, NULL, PJ_ARRAY_SIZE(allowed), allowed);
+
+ // Register "application/sdp" in ACCEPT header
+ pjsip_endpt_add_capability(_endpt, &_mod, PJSIP_H_ACCEPT, NULL, 1, &accepted);
+ }
+
+ _debug("sflphone version %s for %s initialized\n", pj_get_version(), PJ_OS_NAME);
+
+ status = pj_thread_create(_pool, "sflphone", &worker_thread,
+ NULL, 0, 0, &_thread);
+
+ /* Done! */
+ return PJ_SUCCESS;
+
+on_error:
+ //sipDestroy();
+ return status;
+}
+
+bool SIPManager::addAccount(AccountID id, pjsip_regc *regc2, const std::string& server, const std::string& user, const std::string& passwd, const int& timeout) {
+ pj_status_t status;
+ AccountID *currentId = new AccountID(id);
+ //pjsip_cred_info cred;
+ char contactTmp[256];
+ pjsip_regc *regc;
+
+ pj_mutex_lock(_mutex);
+ std::string tmp;
+
+ status = pjsip_regc_create(_endpt, (void *) currentId, ®c_cb, ®c);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to create regc.\n");
+ return status;
+ }
+
+ tmp = "sip:" + server;
+ pj_str_t svr = pj_str((char *) tmp.data());
+ tmp = "<sip:" + user + "@" + server + ">";
+ pj_str_t aor = pj_str((char *) tmp.data());
+
+ sprintf(contactTmp, "<sip:%s@%s:%d>", user.data(), _localExternAddress.data(), _localExternPort);
+ pj_str_t contact = pj_str(contactTmp);
+
+ _debug("Get in %s %d %s\n", svr.ptr, svr.slen, aor.ptr);
+ _debug("after here! contact is %s\n", contact.ptr);
+ status = pjsip_regc_init(regc, &svr, &aor, &aor, 1, &contact, 600); //timeout);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to initialize regc. %d\n", status); //, regc->str_srv_url.ptr);
+ return status;
+ }
+ _debug("after here!\n");
+
+ AccBaseInfo *info = new AccBaseInfo();
+
+ pj_bzero(&info->cred, sizeof (info->cred));
+ pj_strdup2(_pool, &info->cred.username, user.data());
+ info->cred.data_type = PJSIP_CRED_DATA_PLAIN_PASSWD;
+ pj_strdup2(_pool, &info->cred.data, passwd.data());
+ pj_strdup2(_pool, &info->cred.realm, "*");
+ pj_strdup2(_pool, &info->cred.scheme, "digest");
+ pjsip_regc_set_credentials(regc, 1, &info->cred);
+ _debug("after here!\n");
+
+ pjsip_tx_data *tdata;
+ status = pjsip_regc_register(regc, PJ_TRUE, &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to register regc.\n");
+ return status;
+ }
+ _debug("after here!\n");
+
+ status = pjsip_regc_send(regc, tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("! SIP Failure: Unable to send regc request.\n");
+ return status;
+ }
+ _debug("after here!\n");
+
+ info->userName = user;
+ info->server = server;
+ info->id = id;
+ pj_strdup2(_pool, &info->contact, contact.ptr);
+ _accBaseInfoList.push_back(info);
+
+ pj_mutex_unlock(_mutex);
+ return true;
+}
+
+pj_str_t SIPManager::buildContact(char *userName) {
+ pj_str_t contact;
+ char tmp[256];
+
+ //FIXME: IPV6 issue!!
+ _debug("In build Contact %s %s %d\n", userName, _localExternAddress.data(), _localExternPort);
+ sprintf(tmp, "<sip:%s@%s:%d>", userName, _localExternAddress.data(), _localExternPort);
+ _debug("get tmp\n");
+ return pj_str(tmp);
+}
+
+pj_status_t SIPManager::stunServerResolve() {
+ pj_str_t stun_adr;
+ pj_hostent he;
+ pj_stun_config stunCfg;
+ pj_status_t stun_status;
+ pj_sockaddr stun_srv;
+
+ // Initialize STUN configuration
+ pj_stun_config_init(&stunCfg, &_cp.factory, 0, pjsip_endpt_get_ioqueue(_endpt), pjsip_endpt_get_timer_heap(_endpt));
+
+ stun_status = PJ_EPENDING;
+
+ // Init STUN socket
+ stun_adr = pj_str((char *) _stunServer.data());
+ stun_status = pj_sockaddr_in_init(&stun_srv.ipv4, &stun_adr, (pj_uint16_t) 3478);
+
+ if (stun_status != PJ_SUCCESS) {
+ _debug("***********Unresolved stud server***********!\n");
+ stun_status = pj_gethostbyname(&stun_adr, &he);
+
+ if (stun_status == PJ_SUCCESS) {
+ pj_sockaddr_in_init(&stun_srv.ipv4, NULL, 0);
+ stun_srv.ipv4.sin_addr = *(pj_in_addr*) he.h_addr;
+ stun_srv.ipv4.sin_port = pj_htons((pj_uint16_t) 3478);
+ }
+ }
+
+ return stun_status;
+}
+
+int SIPManager::createUDPServer() {
+ pj_status_t status;
+ pj_str_t ipAddr;
+ pj_sockaddr_in bound_addr;
+
+ // Init bound address to ANY
+ pj_memset(&bound_addr, 0, sizeof (bound_addr));
+ bound_addr.sin_addr.s_addr = PJ_INADDR_ANY;
+
+ // Create UDP server socket
+ status = pj_sock_socket(PJ_AF_INET, PJ_SOCK_DGRAM, 0, &_sock);
+ if (status != PJ_SUCCESS) {
+ _debug("(%d) UDP socket() error\n", status);
+ return status;
+ }
+
+ status = pj_sock_bind_in(_sock, pj_ntohl(bound_addr.sin_addr.s_addr), (pj_uint16_t) _localPort);
+ if (status != PJ_SUCCESS) {
+ _debug("(%d) UDP bind() error\n", status);
+ pj_sock_close(_sock);
+ return status;
+ }
+
+ _debug("Use IP: %s\n", _localExternAddress.data());
+
+ // Create UDP-Server (default port: 5060)
+ pjsip_host_port a_name;
+ char tmpIP[32];
+ strcpy(tmpIP, _localExternAddress.data());
+ a_name.host = pj_str(tmpIP);
+ a_name.port = (pj_uint16_t) _localExternPort;
+
+ status = pjsip_udp_transport_attach(_endpt, _sock, &a_name, 1, NULL);
+ if (status != PJ_SUCCESS) {
+ _debug("(%d) Unable to start UDP transport!\n", status);
+ return -1;
+ } else {
+ _debug("UDP server listening on port %d\n", _localExternPort);
+ }
+
+ return 0;
+}
+
+void SIPManager::setStunServer(const char *server) {
+ _stunServer = std::string(server);
+ _useStun = true;
+}
+
+void SIPManager::regc_cb(struct pjsip_regc_cbparam *param) {
+ AccountID *id = static_cast<AccountID *> (param->token);
+ _debug("Account ID is %s, Register result: %d\n", id->data(), param->status);
+ if (param->status == PJ_SUCCESS) {
+ Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::Registered);
+ } else {
+ Manager::instance().getAccountLink(*id)->setRegistrationState(VoIPLink::Error);
+ }
+}
+
+bool
+SIPManager::loadSIPLocalIP() {
+ bool returnValue = true;
+ if (_localIPAddress == "127.0.0.1") {
+ pj_sockaddr ip_addr;
+ if (pj_gethostip(pj_AF_INET(), &ip_addr) != PJ_SUCCESS) {
+ // Update the registration state if no network capabilities found
+ _debug("Get host ip failed!\n");
+ //setRegistrationState( ErrorNetwork );
+ returnValue = false;
+ } else {
+ _localIPAddress = std::string(pj_inet_ntoa(ip_addr.ipv4.sin_addr));
+ _debug(" SIP Info: Checking network, setting local IP address to: %s\n", _localIPAddress.data());
+ }
+ }
+ return returnValue;
+}
+
+/* Worker thread function. */
+int SIPManager::worker_thread(void *arg) {
+ //enum { TIMEOUT = 10 };
+
+ PJ_UNUSED_ARG(arg);
+
+ while (true) {
+ //int count;
+
+ pj_time_val timeout = {0, 10};
+ pjsip_endpt_handle_events(getInstance()->getEndPoint(), &timeout);
+ }
+
+ return 0;
+}
+
+pj_bool_t SIPManager::mod_on_rx_request(pjsip_rx_data *rdata) {
+ pj_status_t status;
+ pj_str_t reason;
+ unsigned options = 0;
+ pjsip_dialog* dialog;
+ pjsip_tx_data *tdata;
+ pjmedia_sdp_session *r_sdp;
+
+ _debug("Callback on_rx_request entered\n");
+
+ PJ_UNUSED_ARG(rdata);
+
+ pjsip_uri *uri = rdata->msg_info.to->uri;
+ pjsip_sip_uri *sip_uri = (pjsip_sip_uri *) pjsip_uri_get_uri(uri);
+
+ std::string userName = std::string(sip_uri->user.ptr, sip_uri->user.slen);
+ std::string server = std::string(sip_uri->host.ptr, sip_uri->host.slen);
+
+ _debug("The receiver is : %s@%s\n", userName.data(), server.data());
+
+ uri = rdata->msg_info.from->uri;
+ sip_uri = (pjsip_sip_uri *) pjsip_uri_get_uri(uri);
+
+ std::string caller = std::string(sip_uri->user.ptr, sip_uri->user.slen);
+ std::string callerServer = std::string(sip_uri->host.ptr, sip_uri->host.slen);
+ std::string peerNumber = caller + "@" + callerServer;
+
+ /* Respond statelessly any non-INVITE requests with 500 */
+ if (rdata->msg_info.msg->line.req.method.id != PJSIP_INVITE_METHOD) {
+ if (rdata->msg_info.msg->line.req.method.id != PJSIP_ACK_METHOD) {
+ reason = pj_str((char*) " user agent unable to handle this request ");
+ pjsip_endpt_respond_stateless(getInstance()->getEndPoint(), rdata, MSG_METHOD_NOT_ALLOWED, &reason, NULL,
+ NULL);
+ return PJ_TRUE;
+ }
+ }
+ // Verify that we can handle the request
+ status = pjsip_inv_verify_request(rdata, &options, NULL, NULL, getInstance()->getEndPoint(), NULL);
+ if (status != PJ_SUCCESS) {
+ reason = pj_str((char*) " user agent unable to handle this INVITE ");
+ pjsip_endpt_respond_stateless(getInstance()->getEndPoint(), rdata, MSG_METHOD_NOT_ALLOWED, &reason, NULL,
+ NULL);
+ return PJ_TRUE;
+ }
+
+ CallID id = Manager::instance().getNewCallID();
+
+ _debug("####################\n##############call id is %s\n", id.c_str());
+ SIPCall* call = new SIPCall(id, Call::Incoming);
+ if (!call) {
+ _debug("! SIP Failure: unable to create an incoming call");
+ return PJ_FALSE;
+ }
+
+ getInstance()->setCallAudioLocal(call);
+ call->setCodecMap(Manager::instance().getCodecDescriptorMap());
+ call->setConnectionState(Call::Progressing);
+ call->setIp(getInstance()->getLocalIP());
+ call->setPeerNumber(peerNumber);
+ if (call->SIPCallInvite(rdata, getInstance()->getAppPool())) {
+ AccountID id = getInstance()->getAccountIdFromNameAndServer(userName, server);
+ _debug("####################\n##############account id is %s\n", id.c_str());
+ if (Manager::instance().incomingCall(call, id)) {
+ Manager::instance().getAccountLink(id)->addCall(call);
+ //addCall(call);
+ _debug("OK\n");
+ } else {
+ delete call;
+ call = 0;
+ }
+ } else {
+ delete call;
+ call = 0;
+ }
+ _debug("Here!!!!!\n");
+
+ /* Create the local dialog (UAS) */
+ status = pjsip_dlg_create_uas(pjsip_ua_instance(), rdata, NULL, &dialog);
+ if (status != PJ_SUCCESS) {
+ pjsip_endpt_respond_stateless(getInstance()->getEndPoint(), rdata, MSG_SERVER_INTERNAL_ERROR, &reason, NULL,
+ NULL);
+ return PJ_TRUE;
+ }
+ // Specify media capability during invite session creation
+ status = pjsip_inv_create_uas(dialog, rdata,
+ call->getLocalSDPSession(), 0, &_invSession);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ _invSession->mod_data[getInstance()->getModId()] = call;
+
+ // Send a 180/Ringing response
+ status = pjsip_inv_initial_answer(_invSession, rdata, PJSIP_SC_RINGING, NULL, NULL, &tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ status = pjsip_inv_send_msg(_invSession, tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ // Update the connection state
+ if (call) {
+ call->setConnectionState(Call::Ringing);
+ }
+
+ /* Done */
+
+ return PJ_SUCCESS;
+}
+
+AccountID SIPManager::getAccountIdFromNameAndServer(const std::string& userName, const std::string& server) {
+ int size = _accBaseInfoList.size();
+ for (int i = 0; i < size; i++) {
+ if (_accBaseInfoList[i]->userName == userName &&
+ _accBaseInfoList[i]->server == server) {
+ _debug("Full match\n");
+ return _accBaseInfoList[i]->id;
+ }
+ }
+
+ for (int i = 0; i < size; i++) {
+ if (_accBaseInfoList[i]->userName == userName) {
+ _debug("Username match\n");
+ return _accBaseInfoList[i]->id;
+ }
+ }
+
+ return AccountNULL;
+}
+
+SIPManager::AccBaseInfo* SIPManager::getAccountInfoFromId(AccountID id) {
+ int size = _accBaseInfoList.size();
+ for (int i = 0; i < size; i++) {
+ if (_accBaseInfoList[i]->id == id) {
+ return _accBaseInfoList[i];
+ }
+ }
+
+ return NULL;
+}
+
+bool
+SIPManager::setCallAudioLocal(SIPCall* call) {
+ // Setting Audio
+ unsigned int callLocalAudioPort = RANDOM_LOCAL_PORT;
+ unsigned int callLocalExternAudioPort = callLocalAudioPort;
+ if (_useStun) {
+ // If use Stun server
+ if (Manager::instance().behindNat(_stunServer, callLocalAudioPort)) {
+ callLocalExternAudioPort = Manager::instance().getFirewallPort();
+ }
+ }
+ _debug(" Setting local audio port to: %d\n", callLocalAudioPort);
+ _debug(" Setting local audio port (external) to: %d\n", callLocalExternAudioPort);
+
+ // Set local audio port for SIPCall(id)
+ call->setLocalIp(_localIPAddress);
+ call->setLocalAudioPort(callLocalAudioPort);
+ call->setLocalExternAudioPort(callLocalExternAudioPort);
+
+ return true;
+}
+
+int SIPManager::answer(SIPCall *call) {
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+
+ if (call->startNegociation(_pool)) {
+ // Create and send a 200(OK) response
+ _debug("Negociation success!\n");
+ status = pjsip_inv_answer(_invSession, PJSIP_SC_OK, NULL, NULL, &tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+ status = pjsip_inv_send_msg(_invSession, tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ return 0;
+ }
+
+ return 1;
+}
+
+bool SIPManager::makeOutgoingCall(const std::string& strTo, SIPCall* call, const AccountID& id) {
+ pj_status_t status;
+ pjsip_dialog *dialog;
+ pjsip_tx_data *tdata;
+ pj_str_t from, to;
+
+ AccBaseInfo* accBase = getAccountInfoFromId(id);
+ std::string strFrom = "sip:" + accBase->userName + "@" + accBase->server;
+
+ _debug("Make a new call from:%s to %s. Contact is %s\n", strFrom.data(), strTo.data(), accBase->contact.ptr);
+
+ from = pj_str((char *) strFrom.data());
+ to = pj_str((char *) strTo.data());
+
+ status = pjsip_dlg_create_uac(pjsip_ua_instance(), &from,
+ &accBase->contact, //NULL,
+ &to,
+ NULL,
+ &dialog);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ setCallAudioLocal(call);
+ call->setIp(getInstance()->getLocalIP());
+
+ // Building the local SDP offer
+ call->createInitialOffer(_pool);
+
+ status = pjsip_inv_create_uac(dialog,
+ call->getLocalSDPSession(), 0, &_invSession);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ pjsip_auth_clt_set_credentials(&dialog->auth_sess, 1, &accBase->cred);
+
+ _invSession->mod_data[_mod.id] = call;
+
+ status = pjsip_inv_invite(_invSession, &tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ status = pjsip_inv_send_msg(_invSession, tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, 1);
+
+ return true;
+}
+
+void SIPManager::call_on_forked(pjsip_inv_session *inv, pjsip_event *e) {
+ //_debug("HERE!\n");
+}
+
+void SIPManager::call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e) {
+ PJ_UNUSED_ARG(inv);
+ _debug("@@@@@@@@@@@@@@@ Here we are, the tsx->state is %d; tsx->role is %d; code is %d; method id is %s.\n",
+ tsx->state, tsx->role, tsx->status_code, tsx->method.name.ptr);
+ if (tsx->state == PJSIP_TSX_STATE_TERMINATED && tsx->role == PJSIP_ROLE_UAC) {
+ int code;
+ pjsip_rx_data *rdata;
+ AccountID accId;
+
+ //Retrieve the body message
+ rdata = e->body.tsx_state.src.rdata;
+
+ code = tsx->status_code;
+ if (code == 200) {
+ _debug("Use this to handle the incoming call!!!\n");
+ SIPCall *call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
+ if(call == NULL)
+ return;
+
+ _debug("The call id is %s\n", call->getCallId().data());
+ //pjsip_rx_data *rdata = e->body.tsx_state.src.rdata;
+ accId = Manager::instance().getAccountFromCall(call->getCallId());
+ SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *>(Manager::instance().getAccountLink(accId));
+ link->SIPCallAnswered(call, rdata);
+ }
+
+ } else if (tsx->state == PJSIP_TSX_STATE_TRYING && tsx->role == PJSIP_ROLE_UAS) {
+ /*_debug("HERE!!!!!!!!!!!!!!!!!!!\n");
+ // Incoming message request
+
+ pjsip_rx_data *rdata;
+ pjsip_msg *msg;
+ pj_status_t status;
+ std::string text;
+
+ // Retrieve the body message
+ rdata = e->body.tsx_state.src.rdata;
+ msg = rdata->msg_info.msg;
+
+ // Respond with OK message
+ status = pjsip_dlg_respond(inv->dlg, rdata, PJSIP_SC_OK, NULL, NULL, NULL);*/
+ } else if (tsx->state == PJSIP_TSX_STATE_PROCEEDING && tsx->role == PJSIP_ROLE_UAC) {
+ int code;
+ pjsip_rx_data *rdata;
+ pjsip_msg *msg;
+ AccountID accId;
+
+ // Retrieve the body message
+ rdata = e->body.tsx_state.src.rdata;
+ msg = rdata->msg_info.msg;
+
+ SIPCall *call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
+ if(call == NULL)
+ return;
+
+ code = msg->line.status.code;
+ switch (code) {
+ case 180:
+ _debug("HERE!!!!!!!!!!!!!!!!!!!\n");
+
+ call->setConnectionState(Call::Ringing);
+ Manager::instance().peerRingingCall(call->getCallId());
+ break;
+ default:
+ break;
+ }
+ } else if (tsx->state == PJSIP_TSX_STATE_COMPLETED && tsx->role == PJSIP_ROLE_UAS) {
+ if(tsx->status_code == 200 && tsx->method.id == PJSIP_BYE_METHOD) {
+ _debug("Receive target phone hang up (bye) message\n");
+ SIPCall *call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
+ if(call == NULL) {
+ _debug("Call has been removed!\n");
+ return;
+ }
+ AccountID accId = Manager::instance().getAccountFromCall(call->getCallId());
+ SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if(link) {
+ link->SIPCallClosed(call);
+ inv->mod_data[getInstance()->getModId()] = NULL;
+ }
+ } else if(tsx->status_code == 200 && tsx->method.id == PJSIP_CANCEL_METHOD) {
+ _debug("Receive cancel message\n");
+ SIPCall *call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
+ if(call == NULL) {
+ _debug("Call has been removed!\n");
+ return;
+ }
+ AccountID accId = Manager::instance().getAccountFromCall(call->getCallId());
+ SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if(link) {
+ link->SIPCallReleased(call);
+ inv->mod_data[getInstance()->getModId()] = NULL;
+ }
+ }
+ } else if (tsx->state == PJSIP_TSX_STATE_COMPLETED && tsx->role == PJSIP_ROLE_UAC) {
+ if (tsx->status_code/100 == 6 || tsx->status_code == 404) {
+ _debug("Receive server error message\n");
+ SIPCall *call = reinterpret_cast<SIPCall *> (inv->mod_data[getInstance()->getModId()]);
+ if(call == NULL) {
+ _debug("Call has been removed!\n");
+ return;
+ }
+ AccountID accId = Manager::instance().getAccountFromCall(call->getCallId());
+ SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ if(link) {
+ link->SIPCallServerFailure(call);
+ inv->mod_data[getInstance()->getModId()] = NULL;
+ }
+ }
+ }
+}
+
+void SIPManager::call_on_state_changed(pjsip_inv_session *inv, pjsip_event *e) {
+ PJ_UNUSED_ARG(inv);
+ /*if( inv->state == PJSIP_INV_STATE_CONFIRMED ) {
+ _debug("Use this to handle the incoming call!!!\n");
+ SIPCall *call = reinterpret_cast<SIPCall *>(inv->mod_data[getInstance()->getModId()]);
+
+
+ _debug("The call id is %s\n", call->getCallId().data());
+ pjsip_rx_data *rdata = e->body.tsx_state.src.rdata;
+ AccountID accId = Manager::instance().getAccountFromCall(call->getCallId());
+ Manager::instance().getAccountLink(accId)->callAnswered(call, rdata);
+ }*/
+}
+
+bool SIPManager::onhold(SIPCall *call) {
+ _debug("*********Before onhold pjsip_inv_reinite begins***********\n");
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+ pjmedia_sdp_attr *attr;
+
+
+ /* Create re-INVITE with new offer */
+ pjmedia_sdp_media_remove_all_attr(call->getLocalSDPSession()->media[0], "sendrecv");
+ attr = pjmedia_sdp_attr_create(_pool, "sendonly", NULL);
+ pjmedia_sdp_media_add_attr(call->getLocalSDPSession()->media[0], attr);
+
+ status = pjsip_inv_reinvite( _invSession, NULL, call->getLocalSDPSession(), &tdata);
+ /* Send the request */
+ status = pjsip_inv_send_msg( _invSession, tdata);
+
+ _debug("*********After pjsip_inv_reinite begins***********\n");
+}
+
+bool SIPManager::offhold(SIPCall *call) {
+ _debug("*********Before offhold pjsip_inv_reinite begins***********\n");
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+ pjmedia_sdp_attr *attr;
+
+
+ /* Create re-INVITE with new offer */
+ pjmedia_sdp_media_remove_all_attr(call->getLocalSDPSession()->media[0], "sendonly");
+ attr = pjmedia_sdp_attr_create(_pool, "sendrecv", NULL);
+ pjmedia_sdp_media_add_attr(call->getLocalSDPSession()->media[0], attr);
+
+ status = pjsip_inv_reinvite( _invSession, NULL, call->getLocalSDPSession(), &tdata);
+ /* Send the request */
+ status = pjsip_inv_send_msg( _invSession, tdata);
+
+ _debug("*********After pjsip_inv_reinite begins***********\n");
+}
+
+bool SIPManager::hangup() {
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+
+ status = pjsip_inv_end_session(_invSession, 404, NULL, &tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
+
+ status = pjsip_inv_send_msg(_invSession, tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
+
+ _invSession->mod_data[getInstance()->getModId()] = NULL;
+ return true;
+}
+
+bool SIPManager::refuse()
+{
+ pj_status_t status;
+ pjsip_tx_data *tdata;
+
+ status = pjsip_inv_end_session(_invSession, PJSIP_SC_DECLINE, NULL, &tdata); //603
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
+
+ status = pjsip_inv_send_msg(_invSession, tdata);
+ PJ_ASSERT_RETURN(status == PJ_SUCCESS, false);
+
+ _invSession->mod_data[getInstance()->getModId()] = NULL;
+ return true;
+}
+
+bool SIPManager::transfer(SIPCall *call, const std::string& to)
+{
+ pjsip_evsub *sub;
+ pjsip_tx_data *tdata;
+ pjsip_dialog *dlg;
+ pjsip_generic_string_hdr *gs_hdr;
+ const pj_str_t str_ref_by = { "Referred-By", 11 };
+ struct pjsip_evsub_user xfer_cb;
+ pj_status_t status;
+ pj_str_t dest;
+
+ pj_strdup2(_pool, &dest, to.data());
+
+ /* Create xfer client subscription. */
+ pj_bzero(&xfer_cb, sizeof(xfer_cb));
+ xfer_cb.on_evsub_state = &xfer_func_cb;
+
+ status = pjsip_xfer_create_uac(_invSession->dlg, &xfer_cb, &sub);
+ if (status != PJ_SUCCESS) {
+ _debug("Unable to create xfer -- %d\n", status);
+ return false;
+ }
+
+ _debug("xfer created!!!!!!!!!!!!!!!\n");
+ /* Associate this call with the client subscription */
+ AccountID accId = Manager::instance().getAccountFromCall(call->getCallId());
+ SIPVoIPLink *link = dynamic_cast<SIPVoIPLink *> (Manager::instance().getAccountLink(accId));
+ pjsip_evsub_set_mod_data(sub, _mod.id, link);
+
+ _debug("call associated!!!!!!!!!!!!!!!\n");
+ /*
+ * Create REFER request.
+ */
+ status = pjsip_xfer_initiate(sub, &dest, &tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("Unable to create REFER request -- %d\n", status);
+ return false;
+ }
+
+ _debug("Here1 !!!!!!!!!!!!!!!!!!!\n");
+ /* Add Referred-By header */
+ //gs_hdr = pjsip_generic_string_hdr_create(tdata->pool, &str_ref_by, &dest);
+
+ //_debug("22222222222222222222222222\n");
+ //pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)gs_hdr);
+ //_debug("3333333333333333333333333333\n");
+
+ /* Send. */
+ status = pjsip_xfer_send_request(sub, tdata);
+ if (status != PJ_SUCCESS) {
+ _debug("Unable to send REFER request -- %d\n", status);
+ return false;
+ }
+
+ return true;
+}
+
+void SIPManager::xfer_func_cb( pjsip_evsub *sub, pjsip_event *event)
+{
+ PJ_UNUSED_ARG(event);
+
+ _debug("#############We are in the callback!!!\n");
+ /*
+ * When subscription is accepted (got 200/OK to REFER), check if
+ * subscription suppressed.
+ */
+ if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACCEPTED) {
+
+ pjsip_rx_data *rdata;
+ pjsip_generic_string_hdr *refer_sub;
+ const pj_str_t REFER_SUB = { "Refer-Sub", 9 };
+
+ SIPVoIPLink *link = reinterpret_cast<SIPVoIPLink *> (pjsip_evsub_get_mod_data(sub,
+ getInstance()->getModId()));
+
+ /* Must be receipt of response message */
+ pj_assert(event->type == PJSIP_EVENT_TSX_STATE &&
+ event->body.tsx_state.type == PJSIP_EVENT_RX_MSG);
+ rdata = event->body.tsx_state.src.rdata;
+
+ /* Find Refer-Sub header */
+ refer_sub = (pjsip_generic_string_hdr*)
+ pjsip_msg_find_hdr_by_name(rdata->msg_info.msg,
+ &REFER_SUB, NULL);
+
+ /* Check if subscription is suppressed */
+ if (refer_sub && pj_stricmp2(&refer_sub->hvalue, "false")==0) {
+ /* Since no subscription is desired, assume that call has been
+ * transfered successfully.
+ */
+ if (link) {
+ link->transferStep2();
+ }
+
+ /* Yes, subscription is suppressed.
+ * Terminate our subscription now.
+ */
+ _debug("Xfer subscription suppressed, terminating event subcription...\n");
+ pjsip_evsub_terminate(sub, PJ_TRUE);
+
+ } else {
+ /* Notify application about call transfer progress.
+ * Initially notify with 100/Accepted status.
+ */
+ _debug("Xfer subscription 100/Accepted received...\n");
+ }
+ }
+ /*
+ * On incoming NOTIFY, notify application about call transfer progress.
+ */
+ else if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_ACTIVE ||
+ pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED)
+ {
+ pjsip_msg *msg;
+ pjsip_msg_body *body;
+ pjsip_status_line status_line;
+ pj_bool_t is_last;
+ pj_bool_t cont;
+ pj_status_t status;
+
+ SIPVoIPLink *link = reinterpret_cast<SIPVoIPLink *> (pjsip_evsub_get_mod_data(sub,
+ getInstance()->getModId()));
+
+ /* When subscription is terminated, clear the xfer_sub member of
+ * the inv_data.
+ */
+ if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) {
+ pjsip_evsub_set_mod_data(sub, getInstance()->getModId(), NULL);
+ _debug("Xfer client subscription terminated\n");
+
+ }
+
+ if (!link || !event) {
+ /* Application is not interested with call progress status */
+ _debug("Either link or event is empty!\n");
+ return;
+ }
+
+ /* This better be a NOTIFY request */
+ if (event->type == PJSIP_EVENT_TSX_STATE &&
+ event->body.tsx_state.type == PJSIP_EVENT_RX_MSG)
+ {
+ pjsip_rx_data *rdata;
+
+ rdata = event->body.tsx_state.src.rdata;
+
+ /* Check if there's body */
+ msg = rdata->msg_info.msg;
+ body = msg->body;
+ if (!body) {
+ _debug("Warning: received NOTIFY without message body\n");
+ return;
+ }
+
+ /* Check for appropriate content */
+ if (pj_stricmp2(&body->content_type.type, "message") != 0 ||
+ pj_stricmp2(&body->content_type.subtype, "sipfrag") != 0)
+ {
+ _debug("Warning: received NOTIFY with non message/sipfrag content\n");
+ return;
+ }
+
+ /* Try to parse the content */
+ status = pjsip_parse_status_line((char*)body->data, body->len,
+ &status_line);
+ if (status != PJ_SUCCESS) {
+ _debug("Warning: received NOTIFY with invalid message/sipfrag content\n");
+ return;
+ }
+
+ } else {
+ _debug("Set code to 500!\n");
+ status_line.code = 500;
+ status_line.reason = *pjsip_get_status_text(500);
+ }
+
+ /* Notify application */
+ is_last = (pjsip_evsub_get_state(sub)==PJSIP_EVSUB_STATE_TERMINATED);
+ cont = !is_last;
+
+ if(status_line.code/100 == 2) {
+ _debug("Try to stop rtp!\n");
+ pjsip_tx_data *tdata;
+
+ status = pjsip_inv_end_session(_invSession, PJSIP_SC_GONE, NULL, &tdata);
+ if(status != PJ_SUCCESS) {
+ _debug("Fail to create end session msg!\n");
+ } else {
+ status = pjsip_inv_send_msg(_invSession, tdata);
+ if(status != PJ_SUCCESS)
+ _debug("Fail to send end session msg!\n");
+ }
+
+ link->transferStep2();
+ cont = PJ_FALSE;
+ }
+
+ if (!cont) {
+ pjsip_evsub_set_mod_data(sub, getInstance()->getModId(), NULL);
+ }
+ }
+
+}
diff --git a/src/sipmanager.h b/src/sipmanager.h
new file mode 100755
index 0000000000..b801f5f05c
--- /dev/null
+++ b/src/sipmanager.h
@@ -0,0 +1,152 @@
+/*
+ * Copyright (C) 2004-2005 Savoir-Faire Linux inc.
+ * Author: Yun Liu <yun.liu@savoirfairelinux.com>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 3 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ */
+
+
+#ifndef _SIPMANAGER_H
+#define _SIPMANAGER_H
+
+#include <pjsip.h>
+#include <pjlib-util.h>
+#include <pjlib.h>
+#include <pjnath/stun_config.h>
+#include <pjsip_simple.h>
+#include <pjsip_ua.h>
+#include <pjmedia/sdp.h>
+#include <pjmedia/sdp_neg.h>
+
+#include <string>
+#include <vector>
+#define MSG_OK 200
+#define MSG_METHOD_NOT_ALLOWED 405
+#define MSG_NOT_ACCEPTABLE_HERE 488
+#define MSG_SERVER_INTERNAL_ERROR 500
+
+typedef std::string AccountID;
+
+class SIPCall;
+
+class SIPManager
+{
+private:
+ /** PJSIP Endpoint */
+ pjsip_endpoint *_endpt;
+ pj_sock_t _sock;
+ //pjsip_module _appMod;
+ pj_caching_pool _cp;
+ pj_pool_t *_pool;
+ pj_thread_t *_voip_listen_thread;
+ pj_mutex_t *_mutex; /** Mutex protection for this data */
+ pjsip_module _mod; /** PJSIP module. */
+ pjsip_module _options_handler;
+ bool _useStun;
+ pj_str_t _stunHost;
+ std::string _stunServer;
+
+ /** Local Extern Address is the IP address seen by peers for SIP listener */
+ std::string _localExternAddress;
+ std::string _localIPAddress;
+
+ /** Local Extern Port is the port seen by peers for SIP listener */
+ unsigned int _localExternPort;
+ unsigned int _localPort;
+
+ pj_thread_t *_thread;
+
+ static SIPManager *_current;
+
+ static pjsip_inv_session* _invSession;
+
+ struct AccBaseInfo {
+ std::string userName;
+ std::string server;
+ AccountID id;
+ pjsip_cred_info cred;
+ pj_str_t contact;
+ };
+
+ typedef std::vector<AccBaseInfo *> AccBaseInfoList;
+ AccBaseInfoList _accBaseInfoList;
+
+public:
+ SIPManager();
+ ~SIPManager();
+
+ pj_status_t sipCreate();
+
+ /**
+ * This method is used to initialize the pjsip
+ */
+ pj_status_t sipInit();
+
+ /** Create SIP UDP Listener */
+ int createUDPServer();
+
+ /** Set whether it will use stun server */
+ void setStunServer(const char *server); //{_stunHost = pj_str(server); _useStun = true;};
+
+ pj_str_t getStunServer() {return _stunHost;}
+
+ //bool addAccount(AccountID id, pjsip_regc *regc, const pj_str_t& registrar, pj_str_t& user, pjsip_cred_info& cred, int& timeout);
+ bool addAccount(AccountID id, pjsip_regc *regc, const std::string& server, const std::string& user, const std::string& passwd, const int& timeout);
+
+ pj_str_t buildContact(char *userName);
+
+ bool loadSIPLocalIP();
+
+ pj_status_t stunServerResolve();
+
+ pjsip_endpoint* getEndPoint() {return _endpt;}
+
+ std::string getLocalIP() {return _localExternAddress;}
+
+ int getModId() {return _mod.id;}
+
+ AccountID getAccountIdFromNameAndServer(const std::string& userName, const std::string& server);
+ AccBaseInfo* getAccountInfoFromId(AccountID id);
+
+ bool setCallAudioLocal(SIPCall* call);
+
+ int answer(SIPCall* call);
+
+ bool hangup();
+
+ bool refuse();
+
+ bool onhold(SIPCall *call);
+ bool offhold(SIPCall *call);
+
+ bool transfer(SIPCall *call, const std::string& to);
+
+ bool makeOutgoingCall(const std::string& to, SIPCall* call, const AccountID& id);
+ pj_pool_t *getAppPool() {return _pool;}
+ static pj_bool_t mod_on_rx_request(pjsip_rx_data *rdata);
+ static pj_bool_t mod_on_rx_response(pjsip_rx_data *rdata) {return PJ_SUCCESS;}
+ static pj_bool_t options_on_rx_request(pjsip_rx_data *rdata) {return PJ_SUCCESS;}
+ static void regc_cb(struct pjsip_regc_cbparam *param);
+ static void xfer_func_cb( pjsip_evsub *sub, pjsip_event *event);
+ static void call_on_media_update( pjsip_inv_session *inv, pj_status_t status) {}
+ static void call_on_state_changed( pjsip_inv_session *inv, pjsip_event *e);
+ static void call_on_forked(pjsip_inv_session *inv, pjsip_event *e);
+ static void call_on_tsx_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e);
+ static int worker_thread(void *arg);
+ static SIPManager* getInstance() {return _current;}
+};
+
+#endif /* _SIPMANAGER_H */
+
--
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