/* $Id$ */ /* * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com) * Copyright (C) 2003-2008 Benny Prijono * * 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 2 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA */ #include #include #define THIS_FILE "pjsua_call.c" /* Retry interval of sending re-INVITE for locking a codec when remote * SDP answer contains multiple codec, in milliseconds. */ #define LOCK_CODEC_RETRY_INTERVAL 200 /* * Max UPDATE/re-INVITE retry to lock codec */ #define LOCK_CODEC_MAX_RETRY 5 /* Determine whether we should restart ICE upon receiving a re-INVITE * with no SDP. */ #define RESTART_ICE_ON_REINVITE 1 /* * The INFO method. */ const pjsip_method pjsip_info_method = { PJSIP_OTHER_METHOD, { "INFO", 4 } }; /* UPDATE method */ static const pjsip_method pjsip_update_method = { PJSIP_OTHER_METHOD, { "UPDATE", 6 } }; /* This callback receives notification from invite session when the * session state has changed. */ static void pjsua_call_on_state_changed(pjsip_inv_session *inv, pjsip_event *e); /* This callback is called by invite session framework when UAC session * has forked. */ static void pjsua_call_on_forked( pjsip_inv_session *inv, pjsip_event *e); /* * Callback to be called when SDP offer/answer negotiation has just completed * in the session. This function will start/update media if negotiation * has succeeded. */ static void pjsua_call_on_media_update(pjsip_inv_session *inv, pj_status_t status); /* * Called when session received new offer. */ static void pjsua_call_on_rx_offer(pjsip_inv_session *inv, struct pjsip_inv_on_rx_offer_cb_param *param); /* * Called when receiving re-INVITE. */ static pj_status_t pjsua_call_on_rx_reinvite(pjsip_inv_session *inv, const pjmedia_sdp_session *offer, pjsip_rx_data *rdata); /* * Called to generate new offer. */ static void pjsua_call_on_create_offer(pjsip_inv_session *inv, pjmedia_sdp_session **offer); /* * This callback is called when transaction state has changed in INVITE * session. We use this to trap: * - incoming REFER request. * - incoming MESSAGE request. */ static void pjsua_call_on_tsx_state_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e); /* * Redirection handler. */ static pjsip_redirect_op pjsua_call_on_redirected(pjsip_inv_session *inv, const pjsip_uri *target, const pjsip_event *e); /* Create SDP for call hold. */ static pj_status_t create_sdp_of_call_hold(pjsua_call *call, pjmedia_sdp_session **p_sdp); /* * Callback called by event framework when the xfer subscription state * has changed. */ static void xfer_client_on_evsub_state( pjsip_evsub *sub, pjsip_event *event); static void xfer_server_on_evsub_state( pjsip_evsub *sub, pjsip_event *event); /* Timer callback to send re-INVITE/UPDATE to lock codec or ICE update */ static void reinv_timer_cb(pj_timer_heap_t *th, pj_timer_entry *entry); /* Check and send reinvite for lock codec and ICE update */ static pj_status_t process_pending_reinvite(pjsua_call *call); /* * Reset call descriptor. */ static void reset_call(pjsua_call_id id) { pjsua_call *call = &pjsua_var.calls[id]; unsigned i; if (call->incoming_data) { pjsip_rx_data_free_cloned(call->incoming_data); call->incoming_data = NULL; } pj_bzero(call, sizeof(*call)); call->index = id; call->last_text.ptr = call->last_text_buf_; call->cname.ptr = call->cname_buf; call->cname.slen = sizeof(call->cname_buf); for (i=0; imedia); ++i) { pjsua_call_media *call_med = &call->media[i]; call_med->ssrc = pj_rand(); call_med->strm.a.conf_slot = PJSUA_INVALID_ID; call_med->strm.v.cap_win_id = PJSUA_INVALID_ID; call_med->strm.v.rdr_win_id = PJSUA_INVALID_ID; call_med->strm.v.strm_dec_slot = PJSUA_INVALID_ID; call_med->strm.v.strm_enc_slot = PJSUA_INVALID_ID; call_med->call = call; call_med->idx = i; call_med->tp_auto_del = PJ_TRUE; } pjsua_call_setting_default(&call->opt); pj_timer_entry_init(&call->reinv_timer, PJ_FALSE, (void*)(pj_size_t)id, &reinv_timer_cb); } /* Get DTMF method type name */ static const char* get_dtmf_method_name(int type) { switch (type) { case PJSUA_DTMF_METHOD_RFC2833: return "RFC2833"; case PJSUA_DTMF_METHOD_SIP_INFO: return "SIP INFO"; } return "(Unknown)"; } /* * Init call subsystem. */ pj_status_t pjsua_call_subsys_init(const pjsua_config *cfg) { pjsip_inv_callback inv_cb; unsigned i; const pj_str_t str_norefersub = { "norefersub", 10 }; pj_status_t status; /* Init calls array. */ for (i=0; i= PJSUA_MAX_CALLS) { pjsua_var.ua_cfg.max_calls = PJSUA_MAX_CALLS; } /* Check the route URI's and force loose route if required */ for (i=0; i= (int)pjsua_var.ua_cfg.max_calls || pjsua_var.next_call_id < 0) { pjsua_var.next_call_id = 0; } for (cid=pjsua_var.next_call_id; cid<(int)pjsua_var.ua_cfg.max_calls; ++cid) { if (pjsua_var.calls[cid].inv == NULL && pjsua_var.calls[cid].async_call.dlg == NULL) { ++pjsua_var.next_call_id; return cid; } } for (cid=0; cid < pjsua_var.next_call_id; ++cid) { if (pjsua_var.calls[cid].inv == NULL && pjsua_var.calls[cid].async_call.dlg == NULL) { ++pjsua_var.next_call_id; return cid; } } #else /* Old algorithm */ for (cid=0; cid<(int)pjsua_var.ua_cfg.max_calls; ++cid) { if (pjsua_var.calls[cid].inv == NULL) return cid; } #endif return PJSUA_INVALID_ID; } /* Get signaling secure level. * Return: * 0: if signaling is not secure * 1: if TLS transport is used for immediate hop * 2: if end-to-end signaling is secure. */ static int get_secure_level(pjsua_acc_id acc_id, const pj_str_t *dst_uri) { const pj_str_t tls = pj_str(";transport=tls"); const pj_str_t sips = pj_str("sips:"); pjsua_acc *acc = &pjsua_var.acc[acc_id]; if (pj_stristr(dst_uri, &sips)) return 2; if (!pj_list_empty(&acc->route_set)) { pjsip_route_hdr *r = acc->route_set.next; pjsip_uri *uri = r->name_addr.uri; pjsip_sip_uri *sip_uri; sip_uri = (pjsip_sip_uri*)pjsip_uri_get_uri(uri); if (pj_stricmp2(&sip_uri->transport_param, "tls")==0) return 1; } else { if (pj_stristr(dst_uri, &tls)) return 1; } return 0; } /* static int call_get_secure_level(pjsua_call *call) { if (call->inv->dlg->secure) return 2; if (!pj_list_empty(&call->inv->dlg->route_set)) { pjsip_route_hdr *r = call->inv->dlg->route_set.next; pjsip_uri *uri = r->name_addr.uri; pjsip_sip_uri *sip_uri; sip_uri = (pjsip_sip_uri*)pjsip_uri_get_uri(uri); if (pj_stricmp2(&sip_uri->transport_param, "tls")==0) return 1; } else { pjsip_sip_uri *sip_uri; if (PJSIP_URI_SCHEME_IS_SIPS(call->inv->dlg->target)) return 2; if (!PJSIP_URI_SCHEME_IS_SIP(call->inv->dlg->target)) return 0; sip_uri = (pjsip_sip_uri*) pjsip_uri_get_uri(call->inv->dlg->target); if (pj_stricmp2(&sip_uri->transport_param, "tls")==0) return 1; } return 0; } */ /* Outgoing call callback when media transport creation is completed. */ static pj_status_t on_make_call_med_tp_complete(pjsua_call_id call_id, const pjsua_med_tp_state_info *info) { pjmedia_sdp_session *offer = NULL; pjsip_inv_session *inv = NULL; pjsua_call *call = &pjsua_var.calls[call_id]; pjsua_acc *acc = &pjsua_var.acc[call->acc_id]; pjsip_dialog *dlg = call->async_call.dlg; unsigned options = 0; pjsip_tx_data *tdata; pj_bool_t cb_called = PJ_FALSE; pj_status_t status = (info? info->status: PJ_SUCCESS); PJSUA_LOCK(); /* Increment the dialog's lock otherwise when invite session creation * fails the dialog will be destroyed prematurely. */ pjsip_dlg_inc_lock(dlg); /* Decrement dialog session. */ pjsip_dlg_dec_session(dlg, &pjsua_var.mod); if (status != PJ_SUCCESS) { pj_str_t err_str; pj_ssize_t title_len; call->last_code = PJSIP_SC_TEMPORARILY_UNAVAILABLE; pj_strcpy2(&call->last_text, "Media init error: "); title_len = call->last_text.slen; err_str = pj_strerror(status, call->last_text_buf_ + title_len, sizeof(call->last_text_buf_) - title_len); call->last_text.slen += err_str.slen; pjsua_perror(THIS_FILE, "Error initializing media channel", status); goto on_error; } /* pjsua_media_channel_deinit() has been called or * call has been hung up. */ if (call->async_call.med_ch_deinit || call->async_call.call_var.out_call.hangup) { PJ_LOG(4,(THIS_FILE, "Call has been hung up or media channel has " "been deinitialized")); goto on_error; } /* Create offer */ if ((call->opt.flag & PJSUA_CALL_NO_SDP_OFFER) == 0) { status = pjsua_media_channel_create_sdp(call->index, dlg->pool, NULL, &offer, NULL); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error initializing media channel", status); goto on_error; } } /* Create the INVITE session: */ options |= PJSIP_INV_SUPPORT_100REL; if (acc->cfg.require_100rel == PJSUA_100REL_MANDATORY) options |= PJSIP_INV_REQUIRE_100REL; if (acc->cfg.use_timer != PJSUA_SIP_TIMER_INACTIVE) { options |= PJSIP_INV_SUPPORT_TIMER; if (acc->cfg.use_timer == PJSUA_SIP_TIMER_REQUIRED) options |= PJSIP_INV_REQUIRE_TIMER; else if (acc->cfg.use_timer == PJSUA_SIP_TIMER_ALWAYS) options |= PJSIP_INV_ALWAYS_USE_TIMER; } status = pjsip_inv_create_uac( dlg, offer, options, &inv); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Invite session creation failed", status); goto on_error; } /* Init Session Timers */ status = pjsip_timer_init_session(inv, &acc->cfg.timer_setting); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Session Timer init failed", status); goto on_error; } /* Create and associate our data in the session. */ call->inv = inv; dlg->mod_data[pjsua_var.mod.id] = call; inv->mod_data[pjsua_var.mod.id] = call; /* If account is locked to specific transport, then lock dialog * to this transport too. */ if (acc->cfg.transport_id != PJSUA_INVALID_ID) { pjsip_tpselector tp_sel; pjsua_init_tpselector(acc->cfg.transport_id, &tp_sel); pjsip_dlg_set_transport(dlg, &tp_sel); } /* Set dialog Route-Set: */ if (!pj_list_empty(&acc->route_set)) pjsip_dlg_set_route_set(dlg, &acc->route_set); /* Set credentials: */ if (acc->cred_cnt) { pjsip_auth_clt_set_credentials( &dlg->auth_sess, acc->cred_cnt, acc->cred); } /* Set authentication preference */ pjsip_auth_clt_set_prefs(&dlg->auth_sess, &acc->cfg.auth_pref); /* Create initial INVITE: */ status = pjsip_inv_invite(inv, &tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create initial INVITE request", status); goto on_error; } /* Add additional headers etc */ pjsua_process_msg_data( tdata, call->async_call.call_var.out_call.msg_data); /* Must increment call counter now */ ++pjsua_var.call_cnt; /* Send initial INVITE: */ status = pjsip_inv_send_msg(inv, tdata); if (status != PJ_SUCCESS) { cb_called = PJ_TRUE; /* Upon failure to send first request, the invite * session would have been cleared. */ inv = NULL; goto on_error; } /* Done. */ call->med_ch_cb = NULL; pjsip_dlg_dec_lock(dlg); PJSUA_UNLOCK(); return PJ_SUCCESS; on_error: if (inv == NULL && call_id != -1 && !cb_called && pjsua_var.ua_cfg.cb.on_call_state) { /* Use user event rather than NULL to avoid crash in * unsuspecting app. */ pjsip_event user_event; PJSIP_EVENT_INIT_USER(user_event, 0, 0, 0, 0); (*pjsua_var.ua_cfg.cb.on_call_state)(call_id, &user_event); } if (dlg) { /* This may destroy the dialog */ pjsip_dlg_dec_lock(dlg); } if (inv != NULL) { pjsip_inv_terminate(inv, PJSIP_SC_OK, PJ_FALSE); } if (call_id != -1) { pjsua_media_channel_deinit(call_id); reset_call(call_id); } call->med_ch_cb = NULL; pjsua_check_snd_dev_idle(); PJSUA_UNLOCK(); return status; } /* * Cleanup call setting flag to avoid one time flags, such as * PJSUA_CALL_UNHOLD, PJSUA_CALL_UPDATE_CONTACT, or * PJSUA_CALL_NO_SDP_OFFER, to be sticky (ticket #1793). */ void pjsua_call_cleanup_flag(pjsua_call_setting *opt) { opt->flag &= ~(PJSUA_CALL_UNHOLD | PJSUA_CALL_UPDATE_CONTACT | PJSUA_CALL_NO_SDP_OFFER | PJSUA_CALL_REINIT_MEDIA | PJSUA_CALL_UPDATE_VIA); } /* * Initialize call settings based on account ID. */ PJ_DEF(void) pjsua_call_setting_default(pjsua_call_setting *opt) { pj_assert(opt); pj_bzero(opt, sizeof(*opt)); opt->flag = PJSUA_CALL_INCLUDE_DISABLED_MEDIA; opt->aud_cnt = 1; #if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0) opt->vid_cnt = 1; opt->req_keyframe_method = PJSUA_VID_REQ_KEYFRAME_SIP_INFO | PJSUA_VID_REQ_KEYFRAME_RTCP_PLI; #endif } /* * Initialize pjsua_call_send_dtmf_param default values. */ PJ_DEF(void) pjsua_call_send_dtmf_param_default( pjsua_call_send_dtmf_param *param) { pj_bzero(param, sizeof(*param)); param->duration = PJSUA_CALL_SEND_DTMF_DURATION_DEFAULT; } static pj_status_t apply_call_setting(pjsua_call *call, const pjsua_call_setting *opt, const pjmedia_sdp_session *rem_sdp) { pj_assert(call); if (!opt) { pjsua_call_cleanup_flag(&call->opt); return PJ_SUCCESS; } #if !PJMEDIA_HAS_VIDEO pj_assert(opt->vid_cnt == 0); #endif call->opt = *opt; if (call->opt.flag & PJSUA_CALL_REINIT_MEDIA) { pjsua_media_channel_deinit(call->index); } /* If call is established or media channel hasn't been initialized, * reinit media channel. */ if ((call->inv && call->inv->state == PJSIP_INV_STATE_CONNECTING && call->med_cnt == 0) || (call->inv && call->inv->state == PJSIP_INV_STATE_CONFIRMED) || (call->opt.flag & PJSUA_CALL_REINIT_MEDIA)) { pjsip_role_e role = rem_sdp? PJSIP_ROLE_UAS : PJSIP_ROLE_UAC; pj_status_t status; status = pjsua_media_channel_init(call->index, role, call->secure_level, call->inv->pool_prov, rem_sdp, NULL, PJ_FALSE, NULL); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error re-initializing media channel", status); return status; } } return PJ_SUCCESS; } static void dlg_set_via(pjsip_dialog *dlg, pjsua_acc *acc) { if (acc->cfg.allow_via_rewrite && acc->via_addr.host.slen > 0) { pjsip_dlg_set_via_sent_by(dlg, &acc->via_addr, acc->via_tp); } else if (!pjsua_sip_acc_is_using_stun(acc->index)) { /* Choose local interface to use in Via if acc is not using * STUN. See https://trac.pjsip.org/repos/ticket/1804 */ pjsip_host_port via_addr; const void *via_tp; if (pjsua_acc_get_uac_addr(acc->index, dlg->pool, &acc->cfg.id, &via_addr, NULL, NULL, &via_tp) == PJ_SUCCESS) { pjsip_dlg_set_via_sent_by(dlg, &via_addr, (pjsip_transport*)via_tp); } } } static pj_status_t dlg_set_target(pjsip_dialog *dlg, const pj_str_t *target) { pjsip_uri *target_uri; pj_str_t tmp; pj_status_t status; /* Parse target & verify */ pj_strdup_with_null(dlg->pool, &tmp, target); target_uri = pjsip_parse_uri(dlg->pool, tmp.ptr, tmp.slen, 0); if (!target_uri) { return PJSIP_EINVALIDURI; } if (!PJSIP_URI_SCHEME_IS_SIP(target_uri) && !PJSIP_URI_SCHEME_IS_SIPS(target_uri)) { return PJSIP_EINVALIDSCHEME; } /* Add the new target */ status = pjsip_target_set_add_uri(&dlg->target_set, dlg->pool, target_uri, 0); if (status != PJ_SUCCESS) return status; /* Set it as current target */ status = pjsip_target_set_set_current(&dlg->target_set, pjsip_target_set_get_next(&dlg->target_set)); if (status != PJ_SUCCESS) return status; /* Update dialog target URI */ dlg->target = target_uri; return PJ_SUCCESS; } /* Get account contact for call and update dialog transport */ void call_update_contact(pjsua_call *call, pj_str_t **new_contact) { pjsip_tpselector tp_sel; pjsua_acc *acc = &pjsua_var.acc[call->acc_id]; if (acc->cfg.force_contact.slen) *new_contact = &acc->cfg.force_contact; else if (acc->contact.slen) *new_contact = &acc->contact; else { /* Non-registering account */ pjsip_dialog *dlg = call->inv->dlg; pj_str_t tmp_contact; pj_status_t status; status = pjsua_acc_create_uac_contact(dlg->pool, &tmp_contact, acc->index, &dlg->remote.info_str); if (status == PJ_SUCCESS) { *new_contact = PJ_POOL_ZALLOC_T(dlg->pool, pj_str_t); **new_contact = tmp_contact; } else { PJ_PERROR(3,(THIS_FILE, status, "Call %d: failed creating contact " "for contact update", call->index)); } } /* When contact is changed, the account transport may have been * changed too, so let's update the dialog's transport too. */ pjsua_init_tpselector(acc->cfg.transport_id, &tp_sel); pjsip_dlg_set_transport(call->inv->dlg, &tp_sel); } /* * Make outgoing call to the specified URI using the specified account. */ PJ_DEF(pj_status_t) pjsua_call_make_call(pjsua_acc_id acc_id, const pj_str_t *dest_uri, const pjsua_call_setting *opt, void *user_data, const pjsua_msg_data *msg_data, pjsua_call_id *p_call_id) { pj_pool_t *tmp_pool = NULL; pjsip_dialog *dlg = NULL; pjsua_acc *acc; pjsua_call *call; int call_id = -1; pj_str_t contact; pj_status_t status; /* Check that account is valid */ PJ_ASSERT_RETURN(acc_id>=0 || acc_id<(int)PJ_ARRAY_SIZE(pjsua_var.acc), PJ_EINVAL); /* Check arguments */ PJ_ASSERT_RETURN(dest_uri, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Making call with acc #%d to %.*s", acc_id, (int)dest_uri->slen, dest_uri->ptr)); pj_log_push_indent(); PJSUA_LOCK(); acc = &pjsua_var.acc[acc_id]; if (!acc->valid) { pjsua_perror(THIS_FILE, "Unable to make call because account " "is not valid", PJ_EINVALIDOP); status = PJ_EINVALIDOP; goto on_error; } /* Find free call slot. */ call_id = alloc_call_id(); if (call_id == PJSUA_INVALID_ID) { pjsua_perror(THIS_FILE, "Error making call", PJ_ETOOMANY); status = PJ_ETOOMANY; goto on_error; } /* Clear call descriptor */ reset_call(call_id); call = &pjsua_var.calls[call_id]; /* Associate session with account */ call->acc_id = acc_id; call->call_hold_type = acc->cfg.call_hold_type; /* Generate per-session RTCP CNAME, according to RFC 7022. */ pj_create_random_string(call->cname_buf, call->cname.slen); /* Apply call setting */ status = apply_call_setting(call, opt, NULL); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Failed to apply call setting", status); goto on_error; } /* Create sound port if none is instantiated, to check if sound device * can be used. But only do this with the conference bridge, as with * audio switchboard (i.e. APS-Direct), we can only open the sound * device once the correct format has been known */ if (!pjsua_var.is_mswitch && pjsua_var.snd_port==NULL && pjsua_var.null_snd==NULL && !pjsua_var.no_snd && call->opt.aud_cnt > 0) { status = pjsua_set_snd_dev(pjsua_var.cap_dev, pjsua_var.play_dev); if (status != PJ_SUCCESS) goto on_error; } /* Create temporary pool */ tmp_pool = pjsua_pool_create("tmpcall10", 512, 256); /* Verify that destination URI is valid before calling * pjsua_acc_create_uac_contact, or otherwise there * a misleading "Invalid Contact URI" error will be printed * when pjsua_acc_create_uac_contact() fails. */ if (1) { pjsip_uri *uri; pj_str_t dup; pj_strdup_with_null(tmp_pool, &dup, dest_uri); uri = pjsip_parse_uri(tmp_pool, dup.ptr, dup.slen, 0); if (uri == NULL) { pjsua_perror(THIS_FILE, "Unable to make call", PJSIP_EINVALIDREQURI); status = PJSIP_EINVALIDREQURI; goto on_error; } } /* Mark call start time. */ pj_gettimeofday(&call->start_time); /* Reset first response time */ call->res_time.sec = 0; /* Create suitable Contact header unless a Contact header has been * set in the account. */ if (acc->contact.slen) { contact = acc->contact; } else { status = pjsua_acc_create_uac_contact(tmp_pool, &contact, acc_id, dest_uri); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to generate Contact header", status); goto on_error; } } /* Create outgoing dialog: */ status = pjsip_dlg_create_uac( pjsip_ua_instance(), &acc->cfg.id, &contact, dest_uri, (msg_data && msg_data->target_uri.slen? &msg_data->target_uri: dest_uri), &dlg); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Dialog creation failed", status); goto on_error; } /* Increment the dialog's lock otherwise when invite session creation * fails the dialog will be destroyed prematurely. */ pjsip_dlg_inc_lock(dlg); dlg_set_via(dlg, acc); /* Calculate call's secure level */ call->secure_level = get_secure_level(acc_id, dest_uri); /* Attach user data */ call->user_data = user_data; /* Store variables required for the callback after the async * media transport creation is completed. */ if (msg_data) { call->async_call.call_var.out_call.msg_data = pjsua_msg_data_clone( dlg->pool, msg_data); } call->async_call.dlg = dlg; /* Temporarily increment dialog session. Without this, dialog will be * prematurely destroyed if dec_lock() is called on the dialog before * the invite session is created. */ pjsip_dlg_inc_session(dlg, &pjsua_var.mod); if ((call->opt.flag & PJSUA_CALL_NO_SDP_OFFER) == 0) { /* Init media channel */ status = pjsua_media_channel_init(call->index, PJSIP_ROLE_UAC, call->secure_level, dlg->pool, NULL, NULL, PJ_TRUE, &on_make_call_med_tp_complete); } if (status == PJ_SUCCESS) { status = on_make_call_med_tp_complete(call->index, NULL); if (status != PJ_SUCCESS) goto on_error; } else if (status != PJ_EPENDING) { pjsua_perror(THIS_FILE, "Error initializing media channel", status); pjsip_dlg_dec_session(dlg, &pjsua_var.mod); goto on_error; } /* Done. */ if (p_call_id) *p_call_id = call_id; pjsip_dlg_dec_lock(dlg); pj_pool_release(tmp_pool); PJSUA_UNLOCK(); pj_log_pop_indent(); return PJ_SUCCESS; on_error: if (dlg) { /* This may destroy the dialog */ pjsip_dlg_dec_lock(dlg); } if (call_id != -1) { pjsua_media_channel_deinit(call_id); reset_call(call_id); } pjsua_check_snd_dev_idle(); if (tmp_pool) pj_pool_release(tmp_pool); PJSUA_UNLOCK(); pj_log_pop_indent(); return status; } /* Get the NAT type information in remote's SDP */ static void update_remote_nat_type(pjsua_call *call, const pjmedia_sdp_session *sdp) { const pjmedia_sdp_attr *xnat; xnat = pjmedia_sdp_attr_find2(sdp->attr_count, sdp->attr, "X-nat", NULL); if (xnat) { call->rem_nat_type = (pj_stun_nat_type) (xnat->value.ptr[0] - '0'); } else { call->rem_nat_type = PJ_STUN_NAT_TYPE_UNKNOWN; } PJ_LOG(5,(THIS_FILE, "Call %d: remote NAT type is %d (%s)", call->index, call->rem_nat_type, pj_stun_get_nat_name(call->rem_nat_type))); } static pj_status_t process_incoming_call_replace(pjsua_call *call, pjsip_dialog *replaced_dlg) { pjsip_inv_session *replaced_inv; struct pjsua_call *replaced_call; pjsip_tx_data *tdata = NULL; pj_status_t status = PJ_SUCCESS; /* Get the invite session in the dialog */ replaced_inv = pjsip_dlg_get_inv_session(replaced_dlg); /* Get the replaced call instance */ replaced_call = (pjsua_call*) replaced_dlg->mod_data[pjsua_var.mod.id]; /* Notify application */ if (pjsua_var.ua_cfg.cb.on_call_replaced) pjsua_var.ua_cfg.cb.on_call_replaced(replaced_call->index, call->index); if (replaced_call->inv->state <= PJSIP_INV_STATE_EARLY && replaced_call->inv->role != PJSIP_ROLE_UAC) { if (replaced_call->last_code > 100 && replaced_call->last_code < 200) { pjsip_status_code code = replaced_call->last_code; pj_str_t *text = &replaced_call->last_text; PJ_LOG(4,(THIS_FILE, "Answering replacement call %d with %d/%.*s", call->index, code, text->slen, text->ptr)); /* Answer the new call with last response in the replaced call */ status = pjsip_inv_answer(call->inv, code, text, NULL, &tdata); } } else { PJ_LOG(4,(THIS_FILE, "Answering replacement call %d with 200/OK", call->index)); /* Answer the new call with 200 response */ status = pjsip_inv_answer(call->inv, 200, NULL, NULL, &tdata); } if (status == PJ_SUCCESS && tdata) status = pjsip_inv_send_msg(call->inv, tdata); if (status != PJ_SUCCESS) pjsua_perror(THIS_FILE, "Error answering session", status); /* Note that inv may be invalid if 200/OK has caused error in * starting the media. */ PJ_LOG(4,(THIS_FILE, "Disconnecting replaced call %d", replaced_call->index)); /* Disconnect replaced invite session */ status = pjsip_inv_end_session(replaced_inv, PJSIP_SC_GONE, NULL, &tdata); if (status == PJ_SUCCESS && tdata) status = pjsip_inv_send_msg(replaced_inv, tdata); if (status != PJ_SUCCESS) pjsua_perror(THIS_FILE, "Error terminating session", status); return status; } static void process_pending_call_answer(pjsua_call *call) { struct call_answer *answer, *next; /* No initial answer yet, this function should be called again later */ if (!call->inv->last_answer) return; answer = call->async_call.call_var.inc_call.answers.next; while (answer != &call->async_call.call_var.inc_call.answers) { next = answer->next; pjsua_call_answer2(call->index, answer->opt, answer->code, answer->reason, answer->msg_data); /* Call might have been disconnected if application is answering * with 200/OK and the media failed to start. * See pjsua_call_answer() below. */ if (!call->inv || !call->inv->pool_prov) break; pj_list_erase(answer); answer = next; } } pj_status_t create_temp_sdp(pj_pool_t *pool, const pjmedia_sdp_session *rem_sdp, pjmedia_sdp_session **p_sdp) { const pj_str_t STR_AUDIO = { "audio", 5 }; const pj_str_t STR_VIDEO = { "video", 5 }; const pj_str_t STR_IP6 = { "IP6", 3}; pjmedia_sdp_session *sdp; pj_sockaddr origin; pj_uint16_t tmp_port = 50123; pj_status_t status = PJ_SUCCESS; pj_str_t tmp_st; unsigned i = 0; pj_bool_t sess_use_ipv4 = PJ_TRUE; /* Get one address to use in the origin field */ pj_sockaddr_init(PJ_AF_INET, &origin, pj_strset2(&tmp_st, "127.0.0.1"), 0); /* Create the base (blank) SDP */ status = pjmedia_endpt_create_base_sdp(pjsua_var.med_endpt, pool, NULL, &origin, &sdp); if (status != PJ_SUCCESS) return status; if (rem_sdp->conn && pj_stricmp(&rem_sdp->conn->addr_type, &STR_IP6)==0) { sess_use_ipv4 = PJ_FALSE; } for (; i< rem_sdp->media_count ; ++i) { pjmedia_sdp_media *m = NULL; pjmedia_sock_info sock_info; pj_bool_t med_use_ipv4 = sess_use_ipv4; if (rem_sdp->media[i]->conn && pj_stricmp(&rem_sdp->media[i]->conn->addr_type, &STR_IP6) == 0) { med_use_ipv4 = PJ_FALSE; } pj_sockaddr_init(med_use_ipv4?PJ_AF_INET:PJ_AF_INET6, &sock_info.rtp_addr_name, med_use_ipv4?pj_strset2(&tmp_st, "127.0.0.1"): pj_strset2(&tmp_st, "::1"), rem_sdp->media[i]->desc.port? (tmp_port++):0); pj_sockaddr_init(med_use_ipv4?PJ_AF_INET:PJ_AF_INET6, &sock_info.rtcp_addr_name, med_use_ipv4?pj_strset2(&tmp_st, "127.0.0.1"): pj_strset2(&tmp_st, "::1"), rem_sdp->media[i]->desc.port? (tmp_port++):0); if (pj_stricmp(&rem_sdp->media[i]->desc.media, &STR_AUDIO)==0) { m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media); status = pjmedia_endpt_create_audio_sdp(pjsua_var.med_endpt, pool, &sock_info, 0, &m); if (status != PJ_SUCCESS) return status; } else if (pj_stricmp(&rem_sdp->media[i]->desc.media, &STR_VIDEO)==0) { #if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0) m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media); status = pjmedia_endpt_create_video_sdp(pjsua_var.med_endpt, pool, &sock_info, 0, &m); if (status != PJ_SUCCESS) return status; #else m = pjmedia_sdp_media_clone_deactivate(pool, rem_sdp->media[i]); #endif } else { m = pjmedia_sdp_media_clone_deactivate(pool, rem_sdp->media[i]); } if (status != PJ_SUCCESS) return status; /* Add connection line, if none */ if (m->conn == NULL && sdp->conn == NULL) { m->conn = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_conn); m->conn->net_type = pj_str("IN"); if (med_use_ipv4) { m->conn->addr_type = pj_str("IP4"); m->conn->addr = pj_str("127.0.0.1"); } else { m->conn->addr_type = pj_str("IP6"); m->conn->addr = pj_str("::1"); } } sdp->media[sdp->media_count++] = m; } *p_sdp = sdp; return PJ_SUCCESS; } static pj_status_t verify_request(const pjsua_call *call, pjsip_rx_data *rdata, pj_bool_t use_tmp_sdp, int *sip_err_code, pjsip_tx_data **response) { const pjmedia_sdp_session *offer = NULL; pjmedia_sdp_session *answer; int err_code = 0; pj_status_t status; /* Get remote SDP offer (if any). */ if (call->inv->neg) { pjmedia_sdp_neg_get_neg_remote(call->inv->neg, &offer); } if (use_tmp_sdp) { if (offer == NULL) return PJ_SUCCESS; /* Create temporary SDP to check for codec support and capability * to handle the required SIP extensions. */ status = create_temp_sdp(call->inv->pool_prov, offer, &answer); if (status != PJ_SUCCESS) { err_code = PJSIP_SC_INTERNAL_SERVER_ERROR; pjsua_perror(THIS_FILE, "Error creating SDP answer", status); } } else { status = pjsua_media_channel_create_sdp(call->index, call->async_call.dlg->pool, offer, &answer, sip_err_code); if (status != PJ_SUCCESS) { err_code = *sip_err_code; pjsua_perror(THIS_FILE, "Error creating SDP answer", status); } else { status = pjsip_inv_set_local_sdp(call->inv, answer); if (status != PJ_SUCCESS) { err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE; pjsua_perror(THIS_FILE, "Error setting local SDP", status); } } } if (status == PJ_SUCCESS) { unsigned options = 0; /* Verify that we can handle the request. */ status = pjsip_inv_verify_request3(rdata, call->inv->pool_prov, &options, offer, answer, NULL, pjsua_var.endpt, response); if (status != PJ_SUCCESS) { /* * No we can't handle the incoming INVITE request. */ pjsua_perror(THIS_FILE, "Request verification failed", status); if (response) err_code = (*response)->msg->line.status.code; else err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE; } } if (sip_err_code && status != PJ_SUCCESS) *sip_err_code = err_code? err_code:PJSIP_ERRNO_TO_SIP_STATUS(status); return status; } /* Incoming call callback when media transport creation is completed. */ static pj_status_t on_incoming_call_med_tp_complete2(pjsua_call_id call_id, const pjsua_med_tp_state_info *info, pjsip_rx_data *rdata, int *sip_err_code, pjsip_tx_data **tdata) { pjsua_call *call = &pjsua_var.calls[call_id]; pjsip_dialog *dlg = call->async_call.dlg; pj_status_t status = (info? info->status: PJ_SUCCESS); int err_code = (info? info->sip_err_code: 0); pjsip_tx_data *response = NULL; PJSUA_LOCK(); /* Increment the dialog's lock to prevent it to be destroyed prematurely, * such as in case of transport error. */ pjsip_dlg_inc_lock(dlg); /* Decrement dialog session. */ pjsip_dlg_dec_session(dlg, &pjsua_var.mod); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error initializing media channel", status); goto on_return; } /* pjsua_media_channel_deinit() has been called. */ if (call->async_call.med_ch_deinit) { pjsua_media_channel_deinit(call->index); call->med_ch_cb = NULL; pjsip_dlg_dec_lock(dlg); PJSUA_UNLOCK(); return PJ_SUCCESS; } status = verify_request(call, rdata, PJ_FALSE, &err_code, &response); on_return: if (status != PJ_SUCCESS) { if (err_code == 0) err_code = PJSIP_ERRNO_TO_SIP_STATUS(status); if (sip_err_code) *sip_err_code = err_code; /* If the callback is called from pjsua_call_on_incoming(), the * invite's state is PJSIP_INV_STATE_NULL, so the invite session * will be terminated later, otherwise we end the session here. */ if (call->inv->state > PJSIP_INV_STATE_NULL) { pj_status_t status_ = PJ_SUCCESS; if (response == NULL) { status_ = pjsip_inv_end_session(call->inv, err_code, NULL, &response); } if (status_ == PJ_SUCCESS && response) status_ = pjsip_inv_send_msg(call->inv, response); } pjsua_media_channel_deinit(call->index); } /* Set the callback to NULL to indicate that the async operation * has completed. */ call->med_ch_cb = NULL; /* Finish any pending process */ if (status == PJ_SUCCESS) { if (call->async_call.call_var.inc_call.replaced_dlg) { /* Process pending call replace */ pjsip_dialog *replaced_dlg = call->async_call.call_var.inc_call.replaced_dlg; process_incoming_call_replace(call, replaced_dlg); } else { /* Process pending call answers */ process_pending_call_answer(call); } } pjsip_dlg_dec_lock(dlg); if (tdata) *tdata = response; PJSUA_UNLOCK(); return status; } static pj_status_t on_incoming_call_med_tp_complete(pjsua_call_id call_id, const pjsua_med_tp_state_info *info) { return on_incoming_call_med_tp_complete2(call_id, info, NULL, NULL, NULL); } /** * Handle incoming INVITE request. * Called by pjsua_core.c */ pj_bool_t pjsua_call_on_incoming(pjsip_rx_data *rdata) { pj_str_t contact; pjsip_dialog *dlg = pjsip_rdata_get_dlg(rdata); pjsip_dialog *replaced_dlg = NULL; pjsip_transaction *tsx = pjsip_rdata_get_tsx(rdata); pjsip_msg *msg = rdata->msg_info.msg; pjsip_tx_data *response = NULL; unsigned options = 0; pjsip_inv_session *inv = NULL; int acc_id; pjsua_call *call = NULL; int call_id = -1; int sip_err_code = PJSIP_SC_INTERNAL_SERVER_ERROR; pjmedia_sdp_session *offer=NULL; pj_bool_t should_dec_dlg = PJ_FALSE; pj_status_t status; /* Don't want to handle anything but INVITE */ if (msg->line.req.method.id != PJSIP_INVITE_METHOD) return PJ_FALSE; /* Don't want to handle anything that's already associated with * existing dialog or transaction. */ if (dlg || tsx) return PJ_FALSE; /* Don't want to accept the call if shutdown is in progress */ if (pjsua_var.thread_quit_flag) { pjsip_endpt_respond_stateless(pjsua_var.endpt, rdata, PJSIP_SC_TEMPORARILY_UNAVAILABLE, NULL, NULL, NULL); return PJ_TRUE; } PJ_LOG(4,(THIS_FILE, "Incoming %s", rdata->msg_info.info)); pj_log_push_indent(); PJSUA_LOCK(); /* Find free call slot. */ call_id = alloc_call_id(); if (call_id == PJSUA_INVALID_ID) { pjsip_endpt_respond_stateless(pjsua_var.endpt, rdata, PJSIP_SC_BUSY_HERE, NULL, NULL, NULL); PJ_LOG(2,(THIS_FILE, "Unable to accept incoming call (too many calls)")); goto on_return; } /* Clear call descriptor */ reset_call(call_id); call = &pjsua_var.calls[call_id]; /* Generate per-session RTCP CNAME, according to RFC 7022. */ pj_create_random_string(call->cname_buf, call->cname.slen); /* Mark call start time. */ pj_gettimeofday(&call->start_time); /* Check INVITE request for Replaces header. If Replaces header is * present, the function will make sure that we can handle the request. */ status = pjsip_replaces_verify_request(rdata, &replaced_dlg, PJ_FALSE, &response); if (status != PJ_SUCCESS) { /* * Something wrong with the Replaces header. */ if (response) { pjsip_response_addr res_addr; pjsip_get_response_addr(response->pool, rdata, &res_addr); pjsip_endpt_send_response(pjsua_var.endpt, &res_addr, response, NULL, NULL); } else { /* Respond with 500 (Internal Server Error) */ pjsip_endpt_respond_stateless(pjsua_var.endpt, rdata, 500, NULL, NULL, NULL); } goto on_return; } /* If this INVITE request contains Replaces header, notify application * about the request so that application can do subsequent checking * if it wants to. */ if (replaced_dlg != NULL && (pjsua_var.ua_cfg.cb.on_call_replace_request || pjsua_var.ua_cfg.cb.on_call_replace_request2)) { pjsua_call *replaced_call; int st_code = 200; pj_str_t st_text = { "OK", 2 }; /* Get the replaced call instance */ replaced_call = (pjsua_call*) replaced_dlg->mod_data[pjsua_var.mod.id]; /* Copy call setting from the replaced call */ call->opt = replaced_call->opt; pjsua_call_cleanup_flag(&call->opt); /* Notify application */ if (pjsua_var.ua_cfg.cb.on_call_replace_request) { pjsua_var.ua_cfg.cb.on_call_replace_request(replaced_call->index, rdata, &st_code, &st_text); } if (pjsua_var.ua_cfg.cb.on_call_replace_request2) { pjsua_var.ua_cfg.cb.on_call_replace_request2(replaced_call->index, rdata, &st_code, &st_text, &call->opt); } /* Must specify final response */ PJ_ASSERT_ON_FAIL(st_code >= 200, st_code = 200); /* Check if application rejects this request. */ if (st_code >= 300) { if (st_text.slen == 2) st_text = *pjsip_get_status_text(st_code); pjsip_endpt_respond(pjsua_var.endpt, NULL, rdata, st_code, &st_text, NULL, NULL, NULL); goto on_return; } } if (!replaced_dlg) { /* Clone rdata. */ pjsip_rx_data_clone(rdata, 0, &call->incoming_data); } /* * Get which account is most likely to be associated with this incoming * call. We need the account to find which contact URI to put for * the call. */ acc_id = call->acc_id = pjsua_acc_find_for_incoming(rdata); if (acc_id == PJSUA_INVALID_ID) { pjsip_endpt_respond_stateless(pjsua_var.endpt, rdata, PJSIP_SC_TEMPORARILY_UNAVAILABLE, NULL, NULL, NULL); PJ_LOG(2,(THIS_FILE, "Unable to accept incoming call (no available account)")); goto on_return; } call->call_hold_type = pjsua_var.acc[acc_id].cfg.call_hold_type; /* Get call's secure level */ if (PJSIP_URI_SCHEME_IS_SIPS(rdata->msg_info.msg->line.req.uri)) call->secure_level = 2; else if (PJSIP_TRANSPORT_IS_SECURE(rdata->tp_info.transport)) call->secure_level = 1; else call->secure_level = 0; /* Parse SDP from incoming request */ if (rdata->msg_info.msg->body) { pjsip_rdata_sdp_info *sdp_info; sdp_info = pjsip_rdata_get_sdp_info(rdata); offer = sdp_info->sdp; status = sdp_info->sdp_err; if (status==PJ_SUCCESS && sdp_info->sdp==NULL && !PJSIP_INV_ACCEPT_UNKNOWN_BODY) { if (sdp_info->body.ptr == NULL) { status = PJSIP_ERRNO_FROM_SIP_STATUS( PJSIP_SC_UNSUPPORTED_MEDIA_TYPE); } else { status = PJSIP_ERRNO_FROM_SIP_STATUS(PJSIP_SC_NOT_ACCEPTABLE); } } if (status != PJ_SUCCESS) { pjsip_hdr hdr_list; /* Check if body really contains SDP. */ if (sdp_info->body.ptr == NULL) { /* Couldn't find "application/sdp" */ pjsip_accept_hdr *acc; pjsua_perror(THIS_FILE, "Unknown Content-Type in incoming "\ "INVITE", status); /* Add Accept header to response */ acc = pjsip_accept_hdr_create(rdata->tp_info.pool); PJ_ASSERT_RETURN(acc, PJ_ENOMEM); acc->values[acc->count++] = pj_str("application/sdp"); pj_list_init(&hdr_list); pj_list_push_back(&hdr_list, acc); pjsip_endpt_respond(pjsua_var.endpt, NULL, rdata, PJSIP_SC_UNSUPPORTED_MEDIA_TYPE, NULL, &hdr_list, NULL, NULL); } else { const pj_str_t reason = pj_str("Bad SDP"); pjsip_warning_hdr *w; pjsua_perror(THIS_FILE, "Bad SDP in incoming INVITE", status); w = pjsip_warning_hdr_create_from_status(rdata->tp_info.pool, pjsip_endpt_name(pjsua_var.endpt), status); pj_list_init(&hdr_list); pj_list_push_back(&hdr_list, w); pjsip_endpt_respond(pjsua_var.endpt, NULL, rdata, 400, &reason, &hdr_list, NULL, NULL); } goto on_return; } /* Do quick checks on SDP before passing it to transports. More elabore * checks will be done in pjsip_inv_verify_request2() below. */ if ((offer) && (offer->media_count==0)) { const pj_str_t reason = pj_str("Missing media in SDP"); pjsip_endpt_respond(pjsua_var.endpt, NULL, rdata, 400, &reason, NULL, NULL, NULL); goto on_return; } } else { offer = NULL; } /* Verify that we can handle the request. */ options |= PJSIP_INV_SUPPORT_100REL; options |= PJSIP_INV_SUPPORT_TIMER; if (pjsua_var.acc[acc_id].cfg.require_100rel == PJSUA_100REL_MANDATORY) options |= PJSIP_INV_REQUIRE_100REL; if (pjsua_var.acc[acc_id].cfg.ice_cfg.enable_ice) options |= PJSIP_INV_SUPPORT_ICE; if (pjsua_var.acc[acc_id].cfg.use_timer == PJSUA_SIP_TIMER_REQUIRED) options |= PJSIP_INV_REQUIRE_TIMER; else if (pjsua_var.acc[acc_id].cfg.use_timer == PJSUA_SIP_TIMER_ALWAYS) options |= PJSIP_INV_ALWAYS_USE_TIMER; status = pjsip_inv_verify_request2(rdata, &options, offer, NULL, NULL, pjsua_var.endpt, &response); if (status != PJ_SUCCESS) { /* * No we can't handle the incoming INVITE request. */ if (response) { pjsip_response_addr res_addr; pjsip_get_response_addr(response->pool, rdata, &res_addr); pjsip_endpt_send_response(pjsua_var.endpt, &res_addr, response, NULL, NULL); } else { /* Respond with 500 (Internal Server Error) */ pjsip_endpt_respond(pjsua_var.endpt, NULL, rdata, 500, NULL, NULL, NULL, NULL); } goto on_return; } /* Get suitable Contact header */ if (pjsua_var.acc[acc_id].contact.slen) { contact = pjsua_var.acc[acc_id].contact; } else { status = pjsua_acc_create_uas_contact(rdata->tp_info.pool, &contact, acc_id, rdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to generate Contact header", status); pjsip_endpt_respond_stateless(pjsua_var.endpt, rdata, 500, NULL, NULL, NULL); goto on_return; } } /* Create dialog: */ status = pjsip_dlg_create_uas_and_inc_lock( pjsip_ua_instance(), rdata, &contact, &dlg); if (status != PJ_SUCCESS) { pjsip_endpt_respond_stateless(pjsua_var.endpt, rdata, 500, NULL, NULL, NULL); goto on_return; } if (pjsua_var.acc[acc_id].cfg.allow_via_rewrite && pjsua_var.acc[acc_id].via_addr.host.slen > 0) { pjsip_dlg_set_via_sent_by(dlg, &pjsua_var.acc[acc_id].via_addr, pjsua_var.acc[acc_id].via_tp); } else if (!pjsua_sip_acc_is_using_stun(acc_id)) { /* Choose local interface to use in Via if acc is not using * STUN. See https://trac.pjsip.org/repos/ticket/1804 */ char target_buf[PJSIP_MAX_URL_SIZE]; pj_str_t target; pjsip_host_port via_addr; const void *via_tp; target.ptr = target_buf; target.slen = pjsip_uri_print(PJSIP_URI_IN_REQ_URI, dlg->target, target_buf, sizeof(target_buf)); if (target.slen < 0) target.slen = 0; if (pjsua_acc_get_uac_addr(acc_id, dlg->pool, &target, &via_addr, NULL, NULL, &via_tp) == PJ_SUCCESS) { pjsip_dlg_set_via_sent_by(dlg, &via_addr, (pjsip_transport*)via_tp); } } /* Set credentials */ if (pjsua_var.acc[acc_id].cred_cnt) { pjsip_auth_clt_set_credentials(&dlg->auth_sess, pjsua_var.acc[acc_id].cred_cnt, pjsua_var.acc[acc_id].cred); } /* Set preference */ pjsip_auth_clt_set_prefs(&dlg->auth_sess, &pjsua_var.acc[acc_id].cfg.auth_pref); /* Disable Session Timers if not prefered and the incoming INVITE request * did not require it. */ if (pjsua_var.acc[acc_id].cfg.use_timer == PJSUA_SIP_TIMER_INACTIVE && (options & PJSIP_INV_REQUIRE_TIMER) == 0) { options &= ~(PJSIP_INV_SUPPORT_TIMER); } /* If 100rel is optional and UAC supports it, use it. */ if ((options & PJSIP_INV_REQUIRE_100REL)==0 && pjsua_var.acc[acc_id].cfg.require_100rel == PJSUA_100REL_OPTIONAL) { const pj_str_t token = { "100rel", 6}; pjsip_dialog_cap_status cap_status; cap_status = pjsip_dlg_remote_has_cap(dlg, PJSIP_H_SUPPORTED, NULL, &token); if (cap_status == PJSIP_DIALOG_CAP_SUPPORTED) options |= PJSIP_INV_REQUIRE_100REL; } /* Create invite session: */ status = pjsip_inv_create_uas( dlg, rdata, NULL, options, &inv); if (status != PJ_SUCCESS) { pjsip_hdr hdr_list; pjsip_warning_hdr *w; w = pjsip_warning_hdr_create_from_status(dlg->pool, pjsip_endpt_name(pjsua_var.endpt), status); pj_list_init(&hdr_list); pj_list_push_back(&hdr_list, w); pjsip_dlg_respond(dlg, rdata, 500, NULL, &hdr_list, NULL); /* Can't terminate dialog because transaction is in progress. pjsip_dlg_terminate(dlg); */ goto on_return; } /* If account is locked to specific transport, then lock dialog * to this transport too. */ if (pjsua_var.acc[acc_id].cfg.transport_id != PJSUA_INVALID_ID) { pjsip_tpselector tp_sel; pjsua_init_tpselector(pjsua_var.acc[acc_id].cfg.transport_id, &tp_sel); pjsip_dlg_set_transport(dlg, &tp_sel); } /* Create and attach pjsua_var data to the dialog */ call->inv = inv; /* Store variables required for the callback after the async * media transport creation is completed. */ call->async_call.dlg = dlg; pj_list_init(&call->async_call.call_var.inc_call.answers); pjsip_dlg_inc_session(dlg, &pjsua_var.mod); should_dec_dlg = PJ_TRUE; /* Init media channel, only when there is offer or call replace request. * For incoming call without SDP offer, media channel init will be done * in pjsua_call_answer(), see ticket #1526. */ if (offer || replaced_dlg) { /* This is only for initial verification, it will check the SDP for * codec support and the capability to handle the required * SIP extensions. */ status = verify_request(call, rdata, PJ_TRUE, &sip_err_code, &response); if (status != PJ_SUCCESS) { pjsip_dlg_inc_lock(dlg); if (response) { pjsip_dlg_send_response(dlg, call->inv->invite_tsx, response); } else { pjsip_dlg_respond(dlg, rdata, sip_err_code, NULL, NULL, NULL); } if (call->inv && call->inv->dlg) { pjsip_inv_terminate(call->inv, sip_err_code, PJ_FALSE); } pjsip_dlg_dec_lock(dlg); call->inv = NULL; call->async_call.dlg = NULL; goto on_return; } status = pjsua_media_channel_init(call->index, PJSIP_ROLE_UAS, call->secure_level, rdata->tp_info.pool, offer, &sip_err_code, PJ_TRUE, &on_incoming_call_med_tp_complete); if (status == PJ_EPENDING) { /* on_incoming_call_med_tp_complete() will call * pjsip_dlg_dec_session(). */ should_dec_dlg = PJ_FALSE; } else if (status == PJ_SUCCESS) { /* on_incoming_call_med_tp_complete2() will call * pjsip_dlg_dec_session(). */ should_dec_dlg = PJ_FALSE; status = on_incoming_call_med_tp_complete2(call_id, NULL, rdata, &sip_err_code, &response); if (status != PJ_SUCCESS) { /* Since the call invite's state is still PJSIP_INV_STATE_NULL, * the invite session was not ended in * on_incoming_call_med_tp_complete(), so we need to send * a response message and terminate the invite here. */ pjsip_dlg_inc_lock(dlg); if (response) { pjsip_dlg_send_response(dlg, call->inv->invite_tsx, response); } else { pjsip_dlg_respond(dlg, rdata, sip_err_code, NULL, NULL, NULL); } if (call->inv && call->inv->dlg) { pjsip_inv_terminate(call->inv, sip_err_code, PJ_FALSE); } pjsip_dlg_dec_lock(dlg); call->inv = NULL; call->async_call.dlg = NULL; goto on_return; } } else if (status != PJ_EPENDING) { pjsua_perror(THIS_FILE, "Error initializing media channel", status); pjsip_dlg_inc_lock(dlg); pjsip_dlg_respond(dlg, rdata, sip_err_code, NULL, NULL, NULL); if (call->inv && call->inv->dlg) { pjsip_inv_terminate(call->inv, sip_err_code, PJ_FALSE); } pjsip_dlg_dec_lock(dlg); call->inv = NULL; call->async_call.dlg = NULL; goto on_return; } } /* Create answer */ /* status = pjsua_media_channel_create_sdp(call->index, rdata->tp_info.pool, offer, &answer, &sip_err_code); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error creating SDP answer", status); pjsip_endpt_respond(pjsua_var.endpt, NULL, rdata, sip_err_code, NULL, NULL, NULL, NULL); goto on_return; } */ /* Init Session Timers */ status = pjsip_timer_init_session(inv, &pjsua_var.acc[acc_id].cfg.timer_setting); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Session Timer init failed", status); pjsip_dlg_respond(dlg, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL, NULL); pjsip_inv_terminate(inv, PJSIP_SC_INTERNAL_SERVER_ERROR, PJ_FALSE); pjsua_media_channel_deinit(call->index); call->inv = NULL; call->async_call.dlg = NULL; goto on_return; } /* Update NAT type of remote endpoint, only when there is SDP in * incoming INVITE! */ if (pjsua_var.ua_cfg.nat_type_in_sdp && inv->neg && pjmedia_sdp_neg_get_state(inv->neg) > PJMEDIA_SDP_NEG_STATE_LOCAL_OFFER) { const pjmedia_sdp_session *remote_sdp; if (pjmedia_sdp_neg_get_neg_remote(inv->neg, &remote_sdp)==PJ_SUCCESS) update_remote_nat_type(call, remote_sdp); } /* Must answer with some response to initial INVITE. We'll do this before * attaching the call to the invite session/dialog, so that the application * will not get notification about this event (on another scenario, it is * also possible that inv_send_msg() fails and causes the invite session to * be disconnected. If we have the call attached at this time, this will * cause the disconnection callback to be called before on_incoming_call() * callback is called, which is not right). */ status = pjsip_inv_initial_answer(inv, rdata, 100, NULL, NULL, &response); if (status != PJ_SUCCESS) { if (response == NULL) { pjsua_perror(THIS_FILE, "Unable to send answer to incoming INVITE", status); pjsip_dlg_respond(dlg, rdata, 500, NULL, NULL, NULL); pjsip_inv_terminate(inv, 500, PJ_FALSE); } else { pjsip_inv_send_msg(inv, response); pjsip_inv_terminate(inv, response->msg->line.status.code, PJ_FALSE); } pjsua_media_channel_deinit(call->index); call->inv = NULL; call->async_call.dlg = NULL; goto on_return; } else { status = pjsip_inv_send_msg(inv, response); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to send 100 response", status); pjsua_media_channel_deinit(call->index); call->inv = NULL; call->async_call.dlg = NULL; goto on_return; } } /* Only do this after sending 100/Trying (really! see the long comment * above) */ if (dlg->mod_data[pjsua_var.mod.id] == NULL) { /* In PJSUA2, on_incoming_call() may be called from * on_media_transport_created() hence this might already set * to allow notification about fail events via on_call_state() and * on_call_tsx_state(). */ dlg->mod_data[pjsua_var.mod.id] = call; inv->mod_data[pjsua_var.mod.id] = call; ++pjsua_var.call_cnt; } /* Check if this request should replace existing call */ if (replaced_dlg) { /* Process call replace. If the media channel init has been completed, * just process now, otherwise, just queue the replaced dialog so * it will be processed once the media channel async init is finished * successfully. */ if (call->med_ch_cb == NULL) { process_incoming_call_replace(call, replaced_dlg); } else { call->async_call.call_var.inc_call.replaced_dlg = replaced_dlg; } } else { /* Notify application if on_incoming_call() is overriden, * otherwise hangup the call with 480 */ if (pjsua_var.ua_cfg.cb.on_incoming_call) { pjsua_var.ua_cfg.cb.on_incoming_call(acc_id, call_id, rdata); /* Notes: * - the call might be reset when it's rejected or hangup * by application from the callback. * - onIncomingCall() may be simulated by onCreateMediaTransport() * when media init is done synchrounously (see #1916). And if app * happens to answer/hangup the call from the callback, the * answer/hangup should have been delayed (see #1923), * so let's process the answer/hangup now. */ if (call->async_call.call_var.inc_call.hangup) { pjsua_call_hangup(call_id, call->last_code, &call->last_text, NULL); } else if (call->med_ch_cb == NULL && call->inv) { process_pending_call_answer(call); } } else { pjsua_call_hangup(call_id, PJSIP_SC_TEMPORARILY_UNAVAILABLE, NULL, NULL); } } /* This INVITE request has been handled. */ on_return: if (dlg) { if (should_dec_dlg) pjsip_dlg_dec_session(dlg, &pjsua_var.mod); pjsip_dlg_dec_lock(dlg); } if (call && call->incoming_data) { pjsip_rx_data_free_cloned(call->incoming_data); call->incoming_data = NULL; } pj_log_pop_indent(); PJSUA_UNLOCK(); return PJ_TRUE; } /* * Check if the specified call has active INVITE session and the INVITE * session has not been disconnected. */ PJ_DEF(pj_bool_t) pjsua_call_is_active(pjsua_call_id call_id) { PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); return pjsua_var.calls[call_id].inv != NULL && pjsua_var.calls[call_id].inv->state != PJSIP_INV_STATE_DISCONNECTED; } /* Acquire lock to the specified call_id */ pj_status_t acquire_call(const char *title, pjsua_call_id call_id, pjsua_call **p_call, pjsip_dialog **p_dlg) { unsigned retry; pjsua_call *call = NULL; pj_bool_t has_pjsua_lock = PJ_FALSE; pj_status_t status = PJ_SUCCESS; pj_time_val time_start, timeout; pjsip_dialog *dlg = NULL; pj_gettimeofday(&time_start); timeout.sec = 0; timeout.msec = PJSUA_ACQUIRE_CALL_TIMEOUT; pj_time_val_normalize(&timeout); for (retry=0; ; ++retry) { if (retry % 10 == 9) { pj_time_val dtime; pj_gettimeofday(&dtime); PJ_TIME_VAL_SUB(dtime, time_start); if (!PJ_TIME_VAL_LT(dtime, timeout)) break; } has_pjsua_lock = PJ_FALSE; status = PJSUA_TRY_LOCK(); if (status != PJ_SUCCESS) { pj_thread_sleep(retry/10); continue; } has_pjsua_lock = PJ_TRUE; call = &pjsua_var.calls[call_id]; if (call->inv) dlg = call->inv->dlg; else dlg = call->async_call.dlg; if (dlg == NULL) { PJSUA_UNLOCK(); PJ_LOG(3,(THIS_FILE, "Invalid call_id %d in %s", call_id, title)); return PJSIP_ESESSIONTERMINATED; } status = pjsip_dlg_try_inc_lock(dlg); if (status != PJ_SUCCESS) { PJSUA_UNLOCK(); pj_thread_sleep(retry/10); continue; } PJSUA_UNLOCK(); break; } if (status != PJ_SUCCESS) { if (has_pjsua_lock == PJ_FALSE) PJ_LOG(1,(THIS_FILE, "Timed-out trying to acquire PJSUA mutex " "(possibly system has deadlocked) in %s", title)); else PJ_LOG(1,(THIS_FILE, "Timed-out trying to acquire dialog mutex " "(possibly system has deadlocked) in %s", title)); return PJ_ETIMEDOUT; } *p_call = call; *p_dlg = dlg; return PJ_SUCCESS; } /* * Obtain detail information about the specified call. */ PJ_DEF(pj_status_t) pjsua_call_get_info( pjsua_call_id call_id, pjsua_call_info *info) { pjsua_call *call; pjsip_dialog *dlg; unsigned mi; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); pj_bzero(info, sizeof(*info)); /* Use PJSUA_LOCK() instead of acquire_call(): * https://trac.pjsip.org/repos/ticket/1371 */ PJSUA_LOCK(); call = &pjsua_var.calls[call_id]; dlg = (call->inv ? call->inv->dlg : call->async_call.dlg); if (!dlg) { PJSUA_UNLOCK(); return PJSIP_ESESSIONTERMINATED; } /* id and role */ info->id = call_id; info->role = dlg->role; info->acc_id = call->acc_id; /* local info */ info->local_info.ptr = info->buf_.local_info; pj_strncpy(&info->local_info, &dlg->local.info_str, sizeof(info->buf_.local_info)); /* local contact */ info->local_contact.ptr = info->buf_.local_contact; info->local_contact.slen = pjsip_uri_print(PJSIP_URI_IN_CONTACT_HDR, dlg->local.contact->uri, info->local_contact.ptr, sizeof(info->buf_.local_contact)); if (info->local_contact.slen < 0) info->local_contact.slen = 0; /* remote info */ info->remote_info.ptr = info->buf_.remote_info; pj_strncpy(&info->remote_info, &dlg->remote.info_str, sizeof(info->buf_.remote_info)); /* remote contact */ if (dlg->remote.contact) { int len; info->remote_contact.ptr = info->buf_.remote_contact; len = pjsip_uri_print(PJSIP_URI_IN_CONTACT_HDR, dlg->remote.contact->uri, info->remote_contact.ptr, sizeof(info->buf_.remote_contact)); if (len < 0) len = 0; info->remote_contact.slen = len; } else { info->remote_contact.slen = 0; } /* call id */ info->call_id.ptr = info->buf_.call_id; pj_strncpy(&info->call_id, &dlg->call_id->id, sizeof(info->buf_.call_id)); /* call setting */ pj_memcpy(&info->setting, &call->opt, sizeof(call->opt)); /* state, state_text */ if (call->inv) { info->state = call->inv->state; if (call->inv->role == PJSIP_ROLE_UAS && info->state == PJSIP_INV_STATE_NULL) { info->state = PJSIP_INV_STATE_INCOMING; } } else if (call->async_call.dlg && call->last_code==0) { info->state = PJSIP_INV_STATE_NULL; } else { info->state = PJSIP_INV_STATE_DISCONNECTED; } info->state_text = pj_str((char*)pjsip_inv_state_name(info->state)); /* If call is disconnected, set the last_status from the cause code */ if (call->inv && call->inv->state >= PJSIP_INV_STATE_DISCONNECTED) { /* last_status, last_status_text */ info->last_status = call->inv->cause; info->last_status_text.ptr = info->buf_.last_status_text; pj_strncpy(&info->last_status_text, &call->inv->cause_text, sizeof(info->buf_.last_status_text)); } else { /* last_status, last_status_text */ info->last_status = call->last_code; info->last_status_text.ptr = info->buf_.last_status_text; pj_strncpy(&info->last_status_text, &call->last_text, sizeof(info->buf_.last_status_text)); } /* Audio & video count offered by remote */ info->rem_offerer = call->rem_offerer; if (call->rem_offerer) { info->rem_aud_cnt = call->rem_aud_cnt; info->rem_vid_cnt = call->rem_vid_cnt; } /* Build array of active media info */ info->media_cnt = 0; for (mi=0; mi < call->med_cnt && info->media_cnt < PJ_ARRAY_SIZE(info->media); ++mi) { pjsua_call_media *call_med = &call->media[mi]; info->media[info->media_cnt].index = mi; info->media[info->media_cnt].status = call_med->state; info->media[info->media_cnt].dir = call_med->dir; info->media[info->media_cnt].type = call_med->type; if (call_med->type == PJMEDIA_TYPE_AUDIO) { info->media[info->media_cnt].stream.aud.conf_slot = call_med->strm.a.conf_slot; } else if (call_med->type == PJMEDIA_TYPE_VIDEO) { pjmedia_vid_dev_index cap_dev = PJMEDIA_VID_INVALID_DEV; info->media[info->media_cnt].stream.vid.win_in = call_med->strm.v.rdr_win_id; info->media[info->media_cnt].stream.vid.dec_slot = call_med->strm.v.strm_dec_slot; info->media[info->media_cnt].stream.vid.enc_slot = call_med->strm.v.strm_enc_slot; if (call_med->strm.v.cap_win_id != PJSUA_INVALID_ID) { cap_dev = call_med->strm.v.cap_dev; } info->media[info->media_cnt].stream.vid.cap_dev = cap_dev; } else { continue; } ++info->media_cnt; } if (call->audio_idx != -1) { info->media_status = call->media[call->audio_idx].state; info->media_dir = call->media[call->audio_idx].dir; info->conf_slot = call->media[call->audio_idx].strm.a.conf_slot; } /* Build array of provisional media info */ info->prov_media_cnt = 0; for (mi=0; mi < call->med_prov_cnt && info->prov_media_cnt < PJ_ARRAY_SIZE(info->prov_media); ++mi) { pjsua_call_media *call_med = &call->media_prov[mi]; info->prov_media[info->prov_media_cnt].index = mi; info->prov_media[info->prov_media_cnt].status = call_med->state; info->prov_media[info->prov_media_cnt].dir = call_med->dir; info->prov_media[info->prov_media_cnt].type = call_med->type; if (call_med->type == PJMEDIA_TYPE_AUDIO) { info->prov_media[info->prov_media_cnt].stream.aud.conf_slot = call_med->strm.a.conf_slot; } else if (call_med->type == PJMEDIA_TYPE_VIDEO) { pjmedia_vid_dev_index cap_dev = PJMEDIA_VID_INVALID_DEV; info->prov_media[info->prov_media_cnt].stream.vid.win_in = call_med->strm.v.rdr_win_id; if (call_med->strm.v.cap_win_id != PJSUA_INVALID_ID) { cap_dev = call_med->strm.v.cap_dev; } info->prov_media[info->prov_media_cnt].stream.vid.cap_dev=cap_dev; } else { continue; } ++info->prov_media_cnt; } /* calculate duration */ if (info->state >= PJSIP_INV_STATE_DISCONNECTED) { info->total_duration = call->dis_time; PJ_TIME_VAL_SUB(info->total_duration, call->start_time); if (call->conn_time.sec) { info->connect_duration = call->dis_time; PJ_TIME_VAL_SUB(info->connect_duration, call->conn_time); } } else if (info->state == PJSIP_INV_STATE_CONFIRMED) { pj_gettimeofday(&info->total_duration); PJ_TIME_VAL_SUB(info->total_duration, call->start_time); pj_gettimeofday(&info->connect_duration); PJ_TIME_VAL_SUB(info->connect_duration, call->conn_time); } else { pj_gettimeofday(&info->total_duration); PJ_TIME_VAL_SUB(info->total_duration, call->start_time); } PJSUA_UNLOCK(); return PJ_SUCCESS; } /* * Check if call remote peer support the specified capability. */ PJ_DEF(pjsip_dialog_cap_status) pjsua_call_remote_has_cap( pjsua_call_id call_id, int htype, const pj_str_t *hname, const pj_str_t *token) { pjsua_call *call; pjsip_dialog *dlg; pj_status_t status; pjsip_dialog_cap_status cap_status; status = acquire_call("pjsua_call_peer_has_cap()", call_id, &call, &dlg); if (status != PJ_SUCCESS) return PJSIP_DIALOG_CAP_UNKNOWN; cap_status = pjsip_dlg_remote_has_cap(dlg, htype, hname, token); pjsip_dlg_dec_lock(dlg); return cap_status; } /* * Attach application specific data to the call. */ PJ_DEF(pj_status_t) pjsua_call_set_user_data( pjsua_call_id call_id, void *user_data) { PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); pjsua_var.calls[call_id].user_data = user_data; return PJ_SUCCESS; } /* * Get user data attached to the call. */ PJ_DEF(void*) pjsua_call_get_user_data(pjsua_call_id call_id) { PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, NULL); return pjsua_var.calls[call_id].user_data; } /* * Get remote's NAT type. */ PJ_DEF(pj_status_t) pjsua_call_get_rem_nat_type(pjsua_call_id call_id, pj_stun_nat_type *p_type) { PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_ASSERT_RETURN(p_type != NULL, PJ_EINVAL); *p_type = pjsua_var.calls[call_id].rem_nat_type; return PJ_SUCCESS; } /* * Get media transport info for the specified media index. */ PJ_DEF(pj_status_t) pjsua_call_get_med_transport_info(pjsua_call_id call_id, unsigned med_idx, pjmedia_transport_info *t) { pjsua_call *call; pjsua_call_media *call_med; pj_status_t status; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_ASSERT_RETURN(t, PJ_EINVAL); PJSUA_LOCK(); call = &pjsua_var.calls[call_id]; if (med_idx >= call->med_cnt) { PJSUA_UNLOCK(); return PJ_EINVAL; } call_med = &call->media[med_idx]; pjmedia_transport_info_init(t); status = pjmedia_transport_get_info(call_med->tp, t); PJSUA_UNLOCK(); return status; } /* Media channel init callback for pjsua_call_answer(). */ static pj_status_t on_answer_call_med_tp_complete(pjsua_call_id call_id, const pjsua_med_tp_state_info *info) { pjsua_call *call = &pjsua_var.calls[call_id]; pjmedia_sdp_session *sdp; int sip_err_code = (info? info->sip_err_code: 0); pj_status_t status = (info? info->status: PJ_SUCCESS); PJSUA_LOCK(); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error initializing media channel", status); goto on_return; } /* pjsua_media_channel_deinit() has been called. */ if (call->async_call.med_ch_deinit) { pjsua_media_channel_deinit(call->index); call->med_ch_cb = NULL; PJSUA_UNLOCK(); return PJ_SUCCESS; } status = pjsua_media_channel_create_sdp(call_id, call->async_call.dlg->pool, NULL, &sdp, &sip_err_code); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error creating SDP answer", status); goto on_return; } status = pjsip_inv_set_local_sdp(call->inv, sdp); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error setting local SDP", status); sip_err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE; goto on_return; } on_return: if (status != PJ_SUCCESS) { /* If the callback is called from pjsua_call_on_incoming(), the * invite's state is PJSIP_INV_STATE_NULL, so the invite session * will be terminated later, otherwise we end the session here. */ if (call->inv->state > PJSIP_INV_STATE_NULL) { pjsip_tx_data *tdata; pj_status_t status_; if (sip_err_code == 0) sip_err_code = PJSIP_ERRNO_TO_SIP_STATUS(status); status_ = pjsip_inv_end_session(call->inv, sip_err_code, NULL, &tdata); if (status_ == PJ_SUCCESS && tdata) status_ = pjsip_inv_send_msg(call->inv, tdata); } pjsua_media_channel_deinit(call->index); } /* Set the callback to NULL to indicate that the async operation * has completed. */ call->med_ch_cb = NULL; /* Finish any pending process */ if (status == PJ_SUCCESS) { /* Process pending call answers */ process_pending_call_answer(call); } PJSUA_UNLOCK(); return status; } /* * Send response to incoming INVITE request. */ PJ_DEF(pj_status_t) pjsua_call_answer( pjsua_call_id call_id, unsigned code, const pj_str_t *reason, const pjsua_msg_data *msg_data) { return pjsua_call_answer2(call_id, NULL, code, reason, msg_data); } /* * Send response to incoming INVITE request. */ PJ_DEF(pj_status_t) pjsua_call_answer2(pjsua_call_id call_id, const pjsua_call_setting *opt, unsigned code, const pj_str_t *reason, const pjsua_msg_data *msg_data) { pjsua_call *call; pjsip_dialog *dlg = NULL; pjsip_tx_data *tdata; pj_status_t status; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Answering call %d: code=%d", call_id, code)); pj_log_push_indent(); status = acquire_call("pjsua_call_answer()", call_id, &call, &dlg); if (status != PJ_SUCCESS) goto on_return; /* Apply call setting, only if status code is 1xx or 2xx. */ if (opt && code < 300) { /* Check if it has not been set previously or it is different to * the previous one. */ if (!call->opt_inited) { call->opt_inited = PJ_TRUE; apply_call_setting(call, opt, NULL); } else if (pj_memcmp(opt, &call->opt, sizeof(*opt)) != 0) { /* Warn application about call setting inconsistency */ PJ_LOG(2,(THIS_FILE, "The call setting changes is ignored.")); } } PJSUA_LOCK(); /* Ticket #1526: When the incoming call contains no SDP offer, the media * channel may have not been initialized at this stage. The media channel * will be initialized here (along with SDP local offer generation) when * the following conditions are met: * - no pending media channel init * - local SDP has not been generated * - call setting has just been set, or SDP offer needs to be sent, i.e: * answer code 183 or 2xx is issued */ if (!call->med_ch_cb && (call->opt_inited || (code==183 || code/100==2)) && (!call->inv->neg || pjmedia_sdp_neg_get_state(call->inv->neg) == PJMEDIA_SDP_NEG_STATE_NULL)) { /* Mark call setting as initialized as it is just about to be used * for initializing the media channel. */ call->opt_inited = PJ_TRUE; status = pjsua_media_channel_init(call->index, PJSIP_ROLE_UAC, call->secure_level, dlg->pool, NULL, NULL, PJ_TRUE, &on_answer_call_med_tp_complete); if (status == PJ_SUCCESS) { status = on_answer_call_med_tp_complete(call->index, NULL); if (status != PJ_SUCCESS) { PJSUA_UNLOCK(); goto on_return; } } else if (status != PJ_EPENDING) { PJSUA_UNLOCK(); pjsua_perror(THIS_FILE, "Error initializing media channel", status); goto on_return; } } /* If media transport creation is not yet completed, we will answer * the call in the media transport creation callback instead. * Or if initial answer is not sent yet, we will answer the call after * initial answer is sent (see #1923). */ if (call->med_ch_cb || !call->inv->last_answer) { struct call_answer *answer; PJ_LOG(4,(THIS_FILE, "Pending answering call %d upon completion " "of media transport", call_id)); answer = PJ_POOL_ZALLOC_T(call->inv->pool_prov, struct call_answer); answer->code = code; if (opt) { answer->opt = PJ_POOL_ZALLOC_T(call->inv->pool_prov, pjsua_call_setting); *answer->opt = *opt; } if (reason) { answer->reason = PJ_POOL_ZALLOC_T(call->inv->pool_prov, pj_str_t); pj_strdup(call->inv->pool_prov, answer->reason, reason); } if (msg_data) { answer->msg_data = pjsua_msg_data_clone(call->inv->pool_prov, msg_data); } pj_list_push_back(&call->async_call.call_var.inc_call.answers, answer); PJSUA_UNLOCK(); if (dlg) pjsip_dlg_dec_lock(dlg); pj_log_pop_indent(); return status; } PJSUA_UNLOCK(); if (call->res_time.sec == 0) pj_gettimeofday(&call->res_time); if (reason && reason->slen == 0) reason = NULL; /* Create response message */ status = pjsip_inv_answer(call->inv, code, reason, NULL, &tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error creating response", status); goto on_return; } /* Call might have been disconnected if application is answering with * 200/OK and the media failed to start. */ if (call->inv == NULL) goto on_return; /* Add additional headers etc */ pjsua_process_msg_data( tdata, msg_data); /* Send the message */ status = pjsip_inv_send_msg(call->inv, tdata); if (status != PJ_SUCCESS) pjsua_perror(THIS_FILE, "Error sending response", status); on_return: if (dlg) pjsip_dlg_dec_lock(dlg); pj_log_pop_indent(); return status; } /* * Send response to incoming INVITE request. */ PJ_DEF(pj_status_t) pjsua_call_answer_with_sdp(pjsua_call_id call_id, const pjmedia_sdp_session *sdp, const pjsua_call_setting *opt, unsigned code, const pj_str_t *reason, const pjsua_msg_data *msg_data) { pjsua_call *call; pjsip_dialog *dlg = NULL; pj_status_t status; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); status = acquire_call("pjsua_call_answer_with_sdp()", call_id, &call, &dlg); if (status != PJ_SUCCESS) return status; status = pjsip_inv_set_sdp_answer(call->inv, sdp); pjsip_dlg_dec_lock(dlg); if (status != PJ_SUCCESS) return status; return pjsua_call_answer2(call_id, opt, code, reason, msg_data); } /* * Hangup call by using method that is appropriate according to the * call state. */ PJ_DEF(pj_status_t) pjsua_call_hangup(pjsua_call_id call_id, unsigned code, const pj_str_t *reason, const pjsua_msg_data *msg_data) { pjsua_call *call; pjsip_dialog *dlg = NULL; pj_status_t status; pjsip_tx_data *tdata; if (call_id<0 || call_id>=(int)pjsua_var.ua_cfg.max_calls) { PJ_LOG(1,(THIS_FILE, "pjsua_call_hangup(): invalid call id %d", call_id)); } PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Call %d hanging up: code=%d..", call_id, code)); pj_log_push_indent(); status = acquire_call("pjsua_call_hangup()", call_id, &call, &dlg); if (status != PJ_SUCCESS) goto on_return; call->hanging_up = PJ_TRUE; /* If media transport creation is not yet completed, we will hangup * the call in the media transport creation callback instead. */ if ((call->med_ch_cb && !call->inv) || ((call->inv != NULL) && (call->inv->state == PJSIP_INV_STATE_NULL))) { PJ_LOG(4,(THIS_FILE, "Pending call %d hangup upon completion " "of media transport", call_id)); if (call->inv && call->inv->role == PJSIP_ROLE_UAS) call->async_call.call_var.inc_call.hangup = PJ_TRUE; else call->async_call.call_var.out_call.hangup = PJ_TRUE; if (code == 0) call->last_code = PJSIP_SC_REQUEST_TERMINATED; else call->last_code = (pjsip_status_code)code; if (reason) { pj_strncpy(&call->last_text, reason, sizeof(call->last_text_buf_)); } goto on_return; } if (code==0) { if (call->inv->state == PJSIP_INV_STATE_CONFIRMED) code = PJSIP_SC_OK; else if (call->inv->role == PJSIP_ROLE_UAS) code = PJSIP_SC_DECLINE; else code = PJSIP_SC_REQUEST_TERMINATED; } status = pjsip_inv_end_session(call->inv, code, reason, &tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Failed to create end session message", status); call->hanging_up = PJ_FALSE; goto on_return; } /* pjsip_inv_end_session may return PJ_SUCCESS with NULL * as p_tdata when INVITE transaction has not been answered * with any provisional responses. */ if (tdata == NULL) goto on_return; /* Add additional headers etc */ pjsua_process_msg_data( tdata, msg_data); /* Send the message */ status = pjsip_inv_send_msg(call->inv, tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Failed to send end session message", status); call->hanging_up = PJ_FALSE; goto on_return; } /* Stop reinvite timer, if it is active */ if (call->reinv_timer.id) { pjsua_cancel_timer(&call->reinv_timer); call->reinv_timer.id = PJ_FALSE; } on_return: if (dlg) pjsip_dlg_dec_lock(dlg); pj_log_pop_indent(); return status; } /* * Accept or reject redirection. */ PJ_DEF(pj_status_t) pjsua_call_process_redirect( pjsua_call_id call_id, pjsip_redirect_op cmd) { pjsua_call *call; pjsip_dialog *dlg; pj_status_t status; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); status = acquire_call("pjsua_call_process_redirect()", call_id, &call, &dlg); if (status != PJ_SUCCESS) return status; status = pjsip_inv_process_redirect(call->inv, cmd, NULL); pjsip_dlg_dec_lock(dlg); return status; } /* * Put the specified call on hold. */ PJ_DEF(pj_status_t) pjsua_call_set_hold(pjsua_call_id call_id, const pjsua_msg_data *msg_data) { return pjsua_call_set_hold2(call_id, 0, msg_data); } PJ_DEF(pj_status_t) pjsua_call_set_hold2(pjsua_call_id call_id, unsigned options, const pjsua_msg_data *msg_data) { pjmedia_sdp_session *sdp; pjsua_call *call; pjsip_dialog *dlg = NULL; pjsip_tx_data *tdata; pj_str_t *new_contact = NULL; pj_status_t status; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Putting call %d on hold", call_id)); pj_log_push_indent(); status = acquire_call("pjsua_call_set_hold()", call_id, &call, &dlg); if (status != PJ_SUCCESS) goto on_return; if (call->inv->state != PJSIP_INV_STATE_CONFIRMED) { PJ_LOG(3,(THIS_FILE, "Can not hold call that is not confirmed")); status = PJSIP_ESESSIONSTATE; goto on_return; } /* We may need to re-initialize media before creating SDP */ if (call->med_prov_cnt == 0) { status = apply_call_setting(call, &call->opt, NULL); if (status != PJ_SUCCESS) goto on_return; } status = create_sdp_of_call_hold(call, &sdp); if (status != PJ_SUCCESS) goto on_return; if ((options & PJSUA_CALL_UPDATE_CONTACT) && pjsua_acc_is_valid(call->acc_id)) { call_update_contact(call, &new_contact); } if ((options & PJSUA_CALL_UPDATE_VIA) && pjsua_acc_is_valid(call->acc_id)) { dlg_set_via(call->inv->dlg, &pjsua_var.acc[call->acc_id]); } if ((call->opt.flag & PJSUA_CALL_UPDATE_TARGET) && msg_data && msg_data->target_uri.slen) { status = dlg_set_target(dlg, &msg_data->target_uri); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to set new target", status); goto on_return; } } /* Create re-INVITE with new offer */ status = pjsip_inv_reinvite( call->inv, new_contact, sdp, &tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create re-INVITE", status); goto on_return; } /* Add additional headers etc */ pjsua_process_msg_data( tdata, msg_data); /* Record the tx_data to keep track the operation */ call->hold_msg = (void*) tdata; /* Send the request */ status = pjsip_inv_send_msg( call->inv, tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to send re-INVITE", status); call->hold_msg = NULL; goto on_return; } /* Set flag that local put the call on hold */ call->local_hold = PJ_TRUE; /* Clear unhold flag */ call->opt.flag &= ~PJSUA_CALL_UNHOLD; on_return: if (dlg) pjsip_dlg_dec_lock(dlg); pj_log_pop_indent(); return status; } /* * Send re-INVITE (to release hold). */ PJ_DEF(pj_status_t) pjsua_call_reinvite( pjsua_call_id call_id, unsigned options, const pjsua_msg_data *msg_data) { pjsua_call *call; pjsip_dialog *dlg = NULL; pj_status_t status; status = acquire_call("pjsua_call_reinvite()", call_id, &call, &dlg); if (status != PJ_SUCCESS) goto on_return; if (options != call->opt.flag) call->opt.flag = options; status = pjsua_call_reinvite2(call_id, &call->opt, msg_data); on_return: if (dlg) pjsip_dlg_dec_lock(dlg); return status; } /* * Send re-INVITE (to release hold). */ PJ_DEF(pj_status_t) pjsua_call_reinvite2(pjsua_call_id call_id, const pjsua_call_setting *opt, const pjsua_msg_data *msg_data) { pjmedia_sdp_session *sdp = NULL; pj_str_t *new_contact = NULL; pjsip_tx_data *tdata; pjsua_call *call; pjsip_dialog *dlg = NULL; pj_status_t status; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Sending re-INVITE on call %d", call_id)); pj_log_push_indent(); status = acquire_call("pjsua_call_reinvite2()", call_id, &call, &dlg); if (status != PJ_SUCCESS) goto on_return; if (pjsua_call_media_is_changing(call)) { PJ_LOG(1,(THIS_FILE, "Unable to reinvite" ERR_MEDIA_CHANGING)); status = PJ_EINVALIDOP; goto on_return; } if (call->inv->state != PJSIP_INV_STATE_CONFIRMED) { PJ_LOG(3,(THIS_FILE, "Can not re-INVITE call that is not confirmed")); status = PJSIP_ESESSIONSTATE; goto on_return; } status = apply_call_setting(call, opt, NULL); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Failed to apply call setting", status); goto on_return; } /* Create SDP */ if (call->local_hold && (call->opt.flag & PJSUA_CALL_UNHOLD)==0) { status = create_sdp_of_call_hold(call, &sdp); } else if ((call->opt.flag & PJSUA_CALL_NO_SDP_OFFER) == 0) { status = pjsua_media_channel_create_sdp(call->index, call->inv->pool_prov, NULL, &sdp, NULL); } if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to get SDP from media endpoint", status); goto on_return; } if ((call->opt.flag & PJSUA_CALL_UPDATE_CONTACT) && pjsua_acc_is_valid(call->acc_id)) { call_update_contact(call, &new_contact); } if ((call->opt.flag & PJSUA_CALL_UPDATE_VIA) && pjsua_acc_is_valid(call->acc_id)) { dlg_set_via(call->inv->dlg, &pjsua_var.acc[call->acc_id]); } if ((call->opt.flag & PJSUA_CALL_UPDATE_TARGET) && msg_data && msg_data->target_uri.slen) { status = dlg_set_target(dlg, &msg_data->target_uri); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to set new target", status); goto on_return; } } /* Create re-INVITE with new offer */ status = pjsip_inv_reinvite( call->inv, new_contact, sdp, &tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create re-INVITE", status); goto on_return; } /* Add additional headers etc */ pjsua_process_msg_data( tdata, msg_data); /* Send the request */ call->med_update_success = PJ_FALSE; status = pjsip_inv_send_msg( call->inv, tdata); if (status == PJ_SUCCESS && ((call->opt.flag & PJSUA_CALL_UNHOLD) && (call->opt.flag & PJSUA_CALL_NO_SDP_OFFER) == 0)) { call->local_hold = PJ_FALSE; } else if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to send re-INVITE", status); goto on_return; } on_return: if (dlg) pjsip_dlg_dec_lock(dlg); pj_log_pop_indent(); return status; } /* * Send UPDATE request. */ PJ_DEF(pj_status_t) pjsua_call_update( pjsua_call_id call_id, unsigned options, const pjsua_msg_data *msg_data) { pjsua_call *call; pjsip_dialog *dlg = NULL; pj_status_t status; status = acquire_call("pjsua_call_update()", call_id, &call, &dlg); if (status != PJ_SUCCESS) goto on_return; if (options != call->opt.flag) call->opt.flag = options; status = pjsua_call_update2(call_id, &call->opt, msg_data); on_return: if (dlg) pjsip_dlg_dec_lock(dlg); return status; } /* * Send UPDATE request. */ PJ_DEF(pj_status_t) pjsua_call_update2(pjsua_call_id call_id, const pjsua_call_setting *opt, const pjsua_msg_data *msg_data) { pjmedia_sdp_session *sdp = NULL; pj_str_t *new_contact = NULL; pjsip_tx_data *tdata; pjsua_call *call; pjsip_dialog *dlg = NULL; pj_status_t status; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Sending UPDATE on call %d", call_id)); pj_log_push_indent(); status = acquire_call("pjsua_call_update2()", call_id, &call, &dlg); if (status != PJ_SUCCESS) goto on_return; /* Don't check media changing if UPDATE is sent without SDP */ if (pjsua_call_media_is_changing(call) && (opt && opt->flag & PJSUA_CALL_NO_SDP_OFFER) == 0) { PJ_LOG(1,(THIS_FILE, "Unable to send UPDATE" ERR_MEDIA_CHANGING)); status = PJ_EINVALIDOP; goto on_return; } status = apply_call_setting(call, opt, NULL); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Failed to apply call setting", status); goto on_return; } /* Create SDP */ if (call->local_hold && (call->opt.flag & PJSUA_CALL_UNHOLD)==0) { status = create_sdp_of_call_hold(call, &sdp); } else if ((call->opt.flag & PJSUA_CALL_NO_SDP_OFFER) == 0) { status = pjsua_media_channel_create_sdp(call->index, call->inv->pool_prov, NULL, &sdp, NULL); } if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to get SDP from media endpoint", status); goto on_return; } if ((call->opt.flag & PJSUA_CALL_UPDATE_CONTACT) && pjsua_acc_is_valid(call->acc_id)) { call_update_contact(call, &new_contact); } if ((call->opt.flag & PJSUA_CALL_UPDATE_VIA) && pjsua_acc_is_valid(call->acc_id)) { dlg_set_via(call->inv->dlg, &pjsua_var.acc[call->acc_id]); } if ((call->opt.flag & PJSUA_CALL_UPDATE_TARGET) && msg_data && msg_data->target_uri.slen) { status = dlg_set_target(dlg, &msg_data->target_uri); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to set new target", status); goto on_return; } } /* Create UPDATE with new offer */ status = pjsip_inv_update(call->inv, new_contact, sdp, &tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create UPDATE request", status); goto on_return; } /* Add additional headers etc */ pjsua_process_msg_data( tdata, msg_data); /* Send the request */ call->med_update_success = PJ_FALSE; status = pjsip_inv_send_msg( call->inv, tdata); if (status == PJ_SUCCESS && ((call->opt.flag & PJSUA_CALL_UNHOLD) && (call->opt.flag & PJSUA_CALL_NO_SDP_OFFER) == 0)) { call->local_hold = PJ_FALSE; } else if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to send UPDATE request", status); goto on_return; } on_return: if (dlg) pjsip_dlg_dec_lock(dlg); pj_log_pop_indent(); return status; } /* * Initiate call transfer to the specified address. */ PJ_DEF(pj_status_t) pjsua_call_xfer( pjsua_call_id call_id, const pj_str_t *dest, const pjsua_msg_data *msg_data) { pjsip_evsub *sub; pjsip_tx_data *tdata; pjsua_call *call; pjsip_dialog *dlg = NULL; 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_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls && dest, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Transferring call %d to %.*s", call_id, (int)dest->slen, dest->ptr)); pj_log_push_indent(); status = acquire_call("pjsua_call_xfer()", call_id, &call, &dlg); if (status != PJ_SUCCESS) goto on_return; /* Create xfer client subscription. */ pj_bzero(&xfer_cb, sizeof(xfer_cb)); xfer_cb.on_evsub_state = &xfer_client_on_evsub_state; status = pjsip_xfer_create_uac(call->inv->dlg, &xfer_cb, &sub); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create xfer", status); goto on_return; } /* Associate this call with the client subscription */ pjsip_evsub_set_mod_data(sub, pjsua_var.mod.id, call); /* * Create REFER request. */ status = pjsip_xfer_initiate(sub, dest, &tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create REFER request", status); goto on_return; } /* Add Referred-By header */ gs_hdr = pjsip_generic_string_hdr_create(tdata->pool, &str_ref_by, &dlg->local.info_str); pjsip_msg_add_hdr(tdata->msg, (pjsip_hdr*)gs_hdr); /* Add additional headers etc */ pjsua_process_msg_data( tdata, msg_data); /* Send. */ status = pjsip_xfer_send_request(sub, tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to send REFER request", status); goto on_return; } /* For simplicity (that's what this program is intended to be!), * leave the original invite session as it is. More advanced application * may want to hold the INVITE, or terminate the invite, or whatever. */ on_return: if (dlg) pjsip_dlg_dec_lock(dlg); pj_log_pop_indent(); return status; } /* * Initiate attended call transfer to the specified address. */ PJ_DEF(pj_status_t) pjsua_call_xfer_replaces( pjsua_call_id call_id, pjsua_call_id dest_call_id, unsigned options, const pjsua_msg_data *msg_data) { pjsua_call *dest_call; pjsip_dialog *dest_dlg; char str_dest_buf[PJSIP_MAX_URL_SIZE*2]; pj_str_t str_dest; int len; char call_id_dest_buf[PJSIP_MAX_URL_SIZE * 2]; int call_id_len; pjsip_uri *uri; pj_status_t status; const pjsip_parser_const_t *pconst; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_ASSERT_RETURN(dest_call_id>=0 && dest_call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Transferring call %d replacing with call %d", call_id, dest_call_id)); pj_log_push_indent(); status = acquire_call("pjsua_call_xfer_replaces()", dest_call_id, &dest_call, &dest_dlg); if (status != PJ_SUCCESS) { pj_log_pop_indent(); return status; } /* * Create REFER destination URI with Replaces field. */ /* Make sure we have sufficient buffer's length */ PJ_ASSERT_ON_FAIL(dest_dlg->remote.info_str.slen + dest_dlg->call_id->id.slen + dest_dlg->remote.info->tag.slen + dest_dlg->local.info->tag.slen + 32 < (long)sizeof(str_dest_buf), { status=PJSIP_EURITOOLONG; goto on_error; }); /* Print URI */ str_dest_buf[0] = '<'; str_dest.slen = 1; uri = (pjsip_uri*) pjsip_uri_get_uri(dest_dlg->remote.info->uri); len = pjsip_uri_print(PJSIP_URI_IN_REQ_URI, uri, str_dest_buf+1, sizeof(str_dest_buf)-1); if (len < 0) { status = PJSIP_EURITOOLONG; goto on_error; } str_dest.slen += len; /* This uses the the same scanner definition used for SIP parsing * to escape the call-id in the refer. * * A common pattern for call-ids is: name@domain. The '@' character, * when used in a URL parameter, throws off many SIP parsers. * URL escape it based off of the allowed characters for header values. */ pconst = pjsip_parser_const(); call_id_len = (int)pj_strncpy2_escape(call_id_dest_buf, &dest_dlg->call_id->id, PJ_ARRAY_SIZE(call_id_dest_buf), &pconst->pjsip_HDR_CHAR_SPEC); if (call_id_len < 0) { status = PJSIP_EURITOOLONG; goto on_error; } /* Build the URI */ len = pj_ansi_snprintf(str_dest_buf + str_dest.slen, sizeof(str_dest_buf) - str_dest.slen, "?%s" "Replaces=%.*s" "%%3Bto-tag%%3D%.*s" "%%3Bfrom-tag%%3D%.*s>", ((options&PJSUA_XFER_NO_REQUIRE_REPLACES) ? "" : "Require=replaces&"), call_id_len, call_id_dest_buf, (int)dest_dlg->remote.info->tag.slen, dest_dlg->remote.info->tag.ptr, (int)dest_dlg->local.info->tag.slen, dest_dlg->local.info->tag.ptr); PJ_ASSERT_ON_FAIL(len > 0 && len <= (int)sizeof(str_dest_buf)-str_dest.slen, { status=PJSIP_EURITOOLONG; goto on_error; }); str_dest.ptr = str_dest_buf; str_dest.slen += len; pjsip_dlg_dec_lock(dest_dlg); status = pjsua_call_xfer(call_id, &str_dest, msg_data); pj_log_pop_indent(); return status; on_error: if (dest_dlg) pjsip_dlg_dec_lock(dest_dlg); pj_log_pop_indent(); return status; } /* * Send DTMF digits to remote. */ PJ_DEF(pj_status_t) pjsua_call_send_dtmf(pjsua_call_id call_id, const pjsua_call_send_dtmf_param *param) { pj_status_t status = PJ_EINVAL; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls && param, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Call %d sending DTMF %.*s using %s method", call_id, (int)param->digits.slen, param->digits.ptr, get_dtmf_method_name(param->method))); if (param->method == PJSUA_DTMF_METHOD_RFC2833) { status = pjsua_call_dial_dtmf(call_id, ¶m->digits); } else if (param->method == PJSUA_DTMF_METHOD_SIP_INFO) { const pj_str_t SIP_INFO = pj_str("INFO"); int i; for (i = 0; i < param->digits.slen; ++i) { char body[80]; pjsua_msg_data msg_data_; pjsua_msg_data_init(&msg_data_); msg_data_.content_type = pj_str("application/dtmf-relay"); pj_ansi_snprintf(body, sizeof(body), "Signal=%c\r\n" "Duration=%d", param->digits.ptr[i], param->duration); msg_data_.msg_body = pj_str(body); status = pjsua_call_send_request(call_id, &SIP_INFO, &msg_data_); } } return status; } /** * Send instant messaging inside INVITE session. */ PJ_DEF(pj_status_t) pjsua_call_send_im( pjsua_call_id call_id, const pj_str_t *mime_type, const pj_str_t *content, const pjsua_msg_data *msg_data, void *user_data) { pjsua_call *call; pjsip_dialog *dlg = NULL; const pj_str_t mime_text_plain = pj_str("text/plain"); pjsip_media_type ctype; pjsua_im_data *im_data; pjsip_tx_data *tdata; pj_status_t status; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Call %d sending %d bytes MESSAGE..", call_id, (int)content->slen)); pj_log_push_indent(); status = acquire_call("pjsua_call_send_im()", call_id, &call, &dlg); if (status != PJ_SUCCESS) goto on_return; /* Set default media type if none is specified */ if (mime_type == NULL) { mime_type = &mime_text_plain; } /* Create request message. */ status = pjsip_dlg_create_request( call->inv->dlg, &pjsip_message_method, -1, &tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create MESSAGE request", status); goto on_return; } /* Add accept header. */ pjsip_msg_add_hdr( tdata->msg, (pjsip_hdr*)pjsua_im_create_accept(tdata->pool)); /* Parse MIME type */ pjsua_parse_media_type(tdata->pool, mime_type, &ctype); /* Create "text/plain" message body. */ tdata->msg->body = pjsip_msg_body_create( tdata->pool, &ctype.type, &ctype.subtype, content); if (tdata->msg->body == NULL) { pjsua_perror(THIS_FILE, "Unable to create msg body", PJ_ENOMEM); pjsip_tx_data_dec_ref(tdata); goto on_return; } /* Add additional headers etc */ pjsua_process_msg_data( tdata, msg_data); /* Create IM data and attach to the request. */ im_data = PJ_POOL_ZALLOC_T(tdata->pool, pjsua_im_data); im_data->acc_id = call->acc_id; im_data->call_id = call_id; im_data->to = call->inv->dlg->remote.info_str; pj_strdup_with_null(tdata->pool, &im_data->body, content); im_data->user_data = user_data; /* Send the request. */ status = pjsip_dlg_send_request( call->inv->dlg, tdata, pjsua_var.mod.id, im_data); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to send MESSAGE request", status); goto on_return; } on_return: if (dlg) pjsip_dlg_dec_lock(dlg); pj_log_pop_indent(); return status; } /* * Send IM typing indication inside INVITE session. */ PJ_DEF(pj_status_t) pjsua_call_send_typing_ind( pjsua_call_id call_id, pj_bool_t is_typing, const pjsua_msg_data*msg_data) { pjsua_call *call; pjsip_dialog *dlg = NULL; pjsip_tx_data *tdata; pj_status_t status; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Call %d sending typing indication..", call_id)); pj_log_push_indent(); status = acquire_call("pjsua_call_send_typing_ind", call_id, &call, &dlg); if (status != PJ_SUCCESS) goto on_return; /* Create request message. */ status = pjsip_dlg_create_request( call->inv->dlg, &pjsip_message_method, -1, &tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create MESSAGE request", status); goto on_return; } /* Create "application/im-iscomposing+xml" msg body. */ tdata->msg->body = pjsip_iscomposing_create_body(tdata->pool, is_typing, NULL, NULL, -1); /* Add additional headers etc */ pjsua_process_msg_data( tdata, msg_data); /* Send the request. */ status = pjsip_dlg_send_request( call->inv->dlg, tdata, -1, NULL); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to send MESSAGE request", status); goto on_return; } on_return: if (dlg) pjsip_dlg_dec_lock(dlg); pj_log_pop_indent(); return status; } /* * Send arbitrary request. */ PJ_DEF(pj_status_t) pjsua_call_send_request(pjsua_call_id call_id, const pj_str_t *method_str, const pjsua_msg_data *msg_data) { pjsua_call *call; pjsip_dialog *dlg = NULL; pjsip_method method; pjsip_tx_data *tdata; pj_status_t status; PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls, PJ_EINVAL); PJ_LOG(4,(THIS_FILE, "Call %d sending %.*s request..", call_id, (int)method_str->slen, method_str->ptr)); pj_log_push_indent(); status = acquire_call("pjsua_call_send_request", call_id, &call, &dlg); if (status != PJ_SUCCESS) goto on_return; /* Init method */ pjsip_method_init_np(&method, (pj_str_t*)method_str); /* Create request message. */ status = pjsip_dlg_create_request( call->inv->dlg, &method, -1, &tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create request", status); goto on_return; } /* Add additional headers etc */ pjsua_process_msg_data( tdata, msg_data); /* Send the request. */ status = pjsip_dlg_send_request( call->inv->dlg, tdata, -1, NULL); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to send request", status); goto on_return; } on_return: if (dlg) pjsip_dlg_dec_lock(dlg); pj_log_pop_indent(); return status; } /* * Terminate all calls. */ PJ_DEF(void) pjsua_call_hangup_all(void) { unsigned i; PJ_LOG(4,(THIS_FILE, "Hangup all calls..")); pj_log_push_indent(); // This may deadlock, see https://trac.pjsip.org/repos/ticket/1305 //PJSUA_LOCK(); for (i=0; iuser_data; pjsip_dialog *dlg; pjsua_call *call; pj_status_t status; PJ_UNUSED_ARG(th); pjsua_var.calls[call_id].reinv_timer.id = PJ_FALSE; pj_log_push_indent(); status = acquire_call("reinv_timer_cb()", call_id, &call, &dlg); if (status != PJ_SUCCESS) { pj_log_pop_indent(); return; } process_pending_reinvite(call); pjsip_dlg_dec_lock(dlg); pj_log_pop_indent(); } /* Check if the specified format can be skipped in counting codecs */ static pj_bool_t is_non_av_fmt(const pjmedia_sdp_media *m, const pj_str_t *fmt) { const pj_str_t STR_TEL = {"telephone-event", 15}; unsigned pt; pt = pj_strtoul(fmt); /* Check for comfort noise */ if (pt == PJMEDIA_RTP_PT_CN) return PJ_TRUE; /* Dynamic PT, check the format name */ if (pt >= 96) { pjmedia_sdp_attr *a; pjmedia_sdp_rtpmap rtpmap; /* Get the format name */ a = pjmedia_sdp_attr_find2(m->attr_count, m->attr, "rtpmap", fmt); if (a && pjmedia_sdp_attr_get_rtpmap(a, &rtpmap)==PJ_SUCCESS) { /* Check for telephone-event */ if (pj_stricmp(&rtpmap.enc_name, &STR_TEL)==0) return PJ_TRUE; } else { /* Invalid SDP, should not reach here */ pj_assert(!"SDP should have been validated!"); return PJ_TRUE; } } return PJ_FALSE; } /* Schedule check for the need of re-INVITE/UPDATE after media update, cases: * - lock codec if remote answerer has given us more than one codecs * - update ICE default transport address if it has changed after ICE * connectivity check. */ void pjsua_call_schedule_reinvite_check(pjsua_call *call, unsigned delay_ms) { pj_time_val delay; /* Stop reinvite timer, if it is active */ if (call->reinv_timer.id) pjsua_cancel_timer(&call->reinv_timer); delay.sec = 0; delay.msec = delay_ms; pj_time_val_normalize(&delay); call->reinv_timer.id = PJ_TRUE; pjsua_schedule_timer(&call->reinv_timer, &delay); } /* Check if lock codec is needed */ static pj_bool_t check_lock_codec(pjsua_call *call) { const pjmedia_sdp_session *local_sdp, *remote_sdp; pj_bool_t has_mult_fmt = PJ_FALSE; unsigned i; pj_status_t status; /* Check if lock codec is disabled */ if (!pjsua_var.acc[call->acc_id].cfg.lock_codec) return PJ_FALSE; /* Check lock codec retry count */ if (call->lock_codec.retry_cnt >= LOCK_CODEC_MAX_RETRY) return PJ_FALSE; /* Check if we are the answerer, we shouldn't need to lock codec */ if (!call->inv->neg || !pjmedia_sdp_neg_was_answer_remote(call->inv->neg)) return PJ_FALSE; /* Check if remote answerer has given us more than one codecs. */ status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &local_sdp); if (status != PJ_SUCCESS) return PJ_FALSE; status = pjmedia_sdp_neg_get_active_remote(call->inv->neg, &remote_sdp); if (status != PJ_SUCCESS) return PJ_FALSE; for (i = 0; i < call->med_cnt && !has_mult_fmt; ++i) { pjsua_call_media *call_med = &call->media[i]; const pjmedia_sdp_media *rem_m, *loc_m; unsigned codec_cnt = 0; unsigned j; /* Skip this if the media is inactive or error */ if (call_med->state == PJSUA_CALL_MEDIA_NONE || call_med->state == PJSUA_CALL_MEDIA_ERROR || call_med->dir == PJMEDIA_DIR_NONE) { continue; } /* Remote may answer with less media lines. */ if (i >= remote_sdp->media_count) continue; rem_m = remote_sdp->media[i]; loc_m = local_sdp->media[i]; /* Verify that media must be active. */ pj_assert(loc_m->desc.port && rem_m->desc.port); PJ_UNUSED_ARG(loc_m); /* Count the formats in the answer. */ for (j=0; jdesc.fmt_count && codec_cnt <= 1; ++j) { if (!is_non_av_fmt(rem_m, &rem_m->desc.fmt[j]) && ++codec_cnt > 1) has_mult_fmt = PJ_TRUE; } } /* Reset retry count when remote answer has one codec */ if (!has_mult_fmt) call->lock_codec.retry_cnt = 0; return has_mult_fmt; } /* Check if ICE setup is complete and if it needs to send reinvite */ static pj_bool_t check_ice_complete(pjsua_call *call, pj_bool_t *need_reinv) { pj_bool_t ice_need_reinv = PJ_FALSE; pj_bool_t ice_complete = PJ_TRUE; unsigned i; /* Check if ICE setup is complete and if it needs reinvite */ for (i = 0; i < call->med_cnt; ++i) { pjsua_call_media *call_med = &call->media[i]; pjmedia_transport_info tpinfo; pjmedia_ice_transport_info *ice_info; if (call_med->tp_st == PJSUA_MED_TP_NULL || call_med->tp_st == PJSUA_MED_TP_DISABLED || call_med->state == PJSUA_CALL_MEDIA_ERROR) { continue; } pjmedia_transport_info_init(&tpinfo); pjmedia_transport_get_info(call_med->tp, &tpinfo); ice_info = (pjmedia_ice_transport_info*) pjmedia_transport_info_get_spc_info( &tpinfo, PJMEDIA_TRANSPORT_TYPE_ICE); /* Check if ICE is active */ if (!ice_info || !ice_info->active) continue; /* Check if ICE setup not completed yet */ if (ice_info->sess_state < PJ_ICE_STRANS_STATE_RUNNING) { ice_complete = PJ_FALSE; break; } /* Check if ICE needs to send reinvite */ if (!ice_need_reinv && ice_info->sess_state == PJ_ICE_STRANS_STATE_RUNNING && ice_info->role == PJ_ICE_SESS_ROLE_CONTROLLING) { pjsua_ice_config *cfg=&pjsua_var.acc[call->acc_id].cfg.ice_cfg; if ((cfg->ice_always_update && !call->reinv_ice_sent) || pj_sockaddr_cmp(&tpinfo.sock_info.rtp_addr_name, &call_med->rtp_addr)) { ice_need_reinv = PJ_TRUE; } } } if (ice_complete && need_reinv) *need_reinv = ice_need_reinv; return ice_complete; } /* Check and send reinvite for lock codec and ICE update */ static pj_status_t process_pending_reinvite(pjsua_call *call) { const pj_str_t ST_UPDATE = {"UPDATE", 6}; pj_pool_t *pool = call->inv->pool_prov; pjsip_inv_session *inv = call->inv; pj_bool_t ice_need_reinv; pj_bool_t ice_completed; pj_bool_t need_lock_codec; pj_bool_t rem_can_update; pjmedia_sdp_session *new_offer; pjsip_tx_data *tdata; unsigned i; pj_status_t status; /* Verify if another SDP negotiation is in progress, e.g: session timer * or another re-INVITE. */ if (inv==NULL || inv->neg==NULL || pjmedia_sdp_neg_get_state(inv->neg)!=PJMEDIA_SDP_NEG_STATE_DONE) { return PJMEDIA_SDPNEG_EINSTATE; } /* Don't do this if call is disconnecting! */ if (inv->state > PJSIP_INV_STATE_CONFIRMED || inv->cause >= 200) { return PJ_EINVALIDOP; } /* Delay this when the SDP negotiation done in call state EARLY and * remote does not support UPDATE method. */ if (inv->state == PJSIP_INV_STATE_EARLY && pjsip_dlg_remote_has_cap(inv->dlg, PJSIP_H_ALLOW, NULL, &ST_UPDATE)!= PJSIP_DIALOG_CAP_SUPPORTED) { call->reinv_pending = PJ_TRUE; return PJ_EPENDING; } /* Check if ICE setup is complete and if it needs reinvite */ ice_completed = check_ice_complete(call, &ice_need_reinv); if (!ice_completed) return PJ_EPENDING; /* Check if we need to lock codec */ need_lock_codec = check_lock_codec(call); /* Check if reinvite is really needed */ if (!need_lock_codec && !ice_need_reinv) return PJ_SUCCESS; /* Okay! So we need to send re-INVITE/UPDATE */ /* Check if remote support UPDATE */ rem_can_update = pjsip_dlg_remote_has_cap(inv->dlg, PJSIP_H_ALLOW, NULL, &ST_UPDATE) == PJSIP_DIALOG_CAP_SUPPORTED; /* Logging stuff */ { const char *ST_ICE_UPDATE = "ICE transport address after " "ICE negotiation"; const char *ST_LOCK_CODEC = "media session to use only one codec"; PJ_LOG(4,(THIS_FILE, "Call %d sending %s for updating %s%s%s", call->index, (rem_can_update? "UPDATE" : "re-INVITE"), (ice_need_reinv? ST_ICE_UPDATE : ST_LOCK_CODEC), (ice_need_reinv && need_lock_codec? " and " : ""), (ice_need_reinv && need_lock_codec? ST_LOCK_CODEC : "") )); } /* Generate SDP re-offer */ status = pjsua_media_channel_create_sdp(call->index, pool, NULL, &new_offer, NULL); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create local SDP", status); return status; } /* Update the new offer so it contains only a codec. Note that * SDP nego has removed unmatched codecs from the offer and the codec * order in the offer has been matched to the answer, so we'll override * the codecs in the just generated SDP with the ones from the active * local SDP and leave just one codec for the next SDP re-offer. */ if (need_lock_codec) { const pjmedia_sdp_session *ref_sdp; /* Get local active SDP as reference */ status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &ref_sdp); if (status != PJ_SUCCESS) return status; /* Verify media count. Note that remote may add/remove media line * in the answer. When answer has less media, it must have been * handled by pjsua_media_channel_update() as disabled media. * When answer has more media, it must have been ignored (treated * as non-exist) anywhere. Local media count should not be updated * at this point, as modifying media count operation (i.e: reinvite, * update, vid_set_strm) is currently blocking, protected with * dialog mutex, and eventually reset SDP nego state to LOCAL OFFER. */ if (call->med_cnt != ref_sdp->media_count || ref_sdp->media_count != new_offer->media_count) { /* Anyway, just in case, let's just return error */ return PJMEDIA_SDPNEG_EINSTATE; } for (i = 0; i < call->med_cnt; ++i) { unsigned j, codec_cnt = 0; const pjmedia_sdp_media *ref_m = ref_sdp->media[i]; pjmedia_sdp_media *m = new_offer->media[i]; pjsua_call_media *call_med = &call->media[i]; /* Verify if media is deactivated */ if (call_med->state == PJSUA_CALL_MEDIA_NONE || call_med->state == PJSUA_CALL_MEDIA_ERROR || call_med->dir == PJMEDIA_DIR_NONE) { continue; } /* Reset formats */ m->desc.fmt_count = 0; pjmedia_sdp_attr_remove_all(&m->attr_count, m->attr, "rtpmap"); pjmedia_sdp_attr_remove_all(&m->attr_count, m->attr, "fmtp"); /* Copy only the first format + any non-AV formats from * the active local SDP. */ for (j = 0; j < ref_m->desc.fmt_count; ++j) { const pj_str_t *fmt = &ref_m->desc.fmt[j]; if (is_non_av_fmt(ref_m, fmt) || (++codec_cnt == 1)) { pjmedia_sdp_attr *a; m->desc.fmt[m->desc.fmt_count++] = *fmt; a = pjmedia_sdp_attr_find2(ref_m->attr_count, ref_m->attr, "rtpmap", fmt); if (a) { pjmedia_sdp_attr_add(&m->attr_count, m->attr, pjmedia_sdp_attr_clone(pool, a)); } a = pjmedia_sdp_attr_find2(ref_m->attr_count, ref_m->attr, "fmtp", fmt); if (a) { pjmedia_sdp_attr_add(&m->attr_count, m->attr, pjmedia_sdp_attr_clone(pool, a)); } } } } } /* Put back original direction and "c=0.0.0.0" line */ { const pjmedia_sdp_session *cur_sdp; /* Get local active SDP */ status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &cur_sdp); if (status != PJ_SUCCESS) return status; /* Make sure media count has not been changed */ if (call->med_cnt != cur_sdp->media_count) return PJMEDIA_SDPNEG_EINSTATE; for (i = 0; i < call->med_cnt; ++i) { const pjmedia_sdp_media *m = cur_sdp->media[i]; pjmedia_sdp_media *new_m = new_offer->media[i]; pjsua_call_media *call_med = &call->media[i]; pjmedia_sdp_attr *a = NULL; /* Update direction to the current dir */ pjmedia_sdp_media_remove_all_attr(new_m, "sendrecv"); pjmedia_sdp_media_remove_all_attr(new_m, "sendonly"); pjmedia_sdp_media_remove_all_attr(new_m, "recvonly"); pjmedia_sdp_media_remove_all_attr(new_m, "inactive"); if (call_med->dir == PJMEDIA_DIR_ENCODING_DECODING) { a = pjmedia_sdp_attr_create(pool, "sendrecv", NULL); } else if (call_med->dir == PJMEDIA_DIR_ENCODING) { a = pjmedia_sdp_attr_create(pool, "sendonly", NULL); } else if (call_med->dir == PJMEDIA_DIR_DECODING) { a = pjmedia_sdp_attr_create(pool, "recvonly", NULL); } else { const pjmedia_sdp_conn *conn; a = pjmedia_sdp_attr_create(pool, "inactive", NULL); /* Also check if the original c= line address is zero */ conn = m->conn; if (!conn) conn = cur_sdp->conn; if (pj_strcmp2(&conn->addr, "0.0.0.0")==0 || pj_strcmp2(&conn->addr, "0")==0) { if (!new_m->conn) { new_m->conn = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_conn); } if (pj_strcmp2(&new_m->conn->addr, "0.0.0.0")) { new_m->conn->net_type = pj_str("IN"); new_m->conn->addr_type = pj_str("IP4"); new_m->conn->addr = pj_str("0.0.0.0"); } } } pj_assert(a); pjmedia_sdp_media_add_attr(new_m, a); } } if (rem_can_update) { status = pjsip_inv_update(inv, NULL, new_offer, &tdata); } else { status = pjsip_inv_reinvite(inv, NULL, new_offer, &tdata); } if (status==PJ_EINVALIDOP && ++call->lock_codec.retry_cnt < LOCK_CODEC_MAX_RETRY) { /* Ups, let's reschedule again */ pjsua_call_schedule_reinvite_check(call, LOCK_CODEC_RETRY_INTERVAL); return PJ_SUCCESS; } else if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error creating UPDATE/re-INVITE", status); return status; } /* Send the UPDATE/re-INVITE request */ status = pjsip_inv_send_msg(inv, tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Error sending UPDATE/re-INVITE", status); return status; } /* Update flags */ if (ice_need_reinv) call->reinv_ice_sent = PJ_TRUE; if (need_lock_codec) ++call->lock_codec.retry_cnt; return PJ_SUCCESS; } /* * This callback receives notification from invite session when the * session state has changed. */ static void pjsua_call_on_state_changed(pjsip_inv_session *inv, pjsip_event *e) { pjsua_call *call; unsigned num_locks = 0; pj_log_push_indent(); call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id]; if (!call) { pj_log_pop_indent(); return; } /* Get call times */ switch (inv->state) { case PJSIP_INV_STATE_EARLY: case PJSIP_INV_STATE_CONNECTING: if (call->res_time.sec == 0) pj_gettimeofday(&call->res_time); call->last_code = (pjsip_status_code) e->body.tsx_state.tsx->status_code; pj_strncpy(&call->last_text, &e->body.tsx_state.tsx->status_text, sizeof(call->last_text_buf_)); break; case PJSIP_INV_STATE_CONFIRMED: pj_gettimeofday(&call->conn_time); /* See if auto reinvite was pended as media update was done in the * EARLY state and remote does not support UPDATE. */ if (call->reinv_pending) { call->reinv_pending = PJ_FALSE; pjsua_call_schedule_reinvite_check(call, 0); } break; case PJSIP_INV_STATE_DISCONNECTED: pj_gettimeofday(&call->dis_time); if (call->res_time.sec == 0) pj_gettimeofday(&call->res_time); if (e->type == PJSIP_EVENT_TSX_STATE && e->body.tsx_state.tsx->status_code > call->last_code) { call->last_code = (pjsip_status_code) e->body.tsx_state.tsx->status_code; pj_strncpy(&call->last_text, &e->body.tsx_state.tsx->status_text, sizeof(call->last_text_buf_)); } else { call->last_code = PJSIP_SC_REQUEST_TERMINATED; pj_strncpy(&call->last_text, pjsip_get_status_text(call->last_code), sizeof(call->last_text_buf_)); } /* Stop reinvite timer, if it is active */ if (call->reinv_timer.id) { pjsua_cancel_timer(&call->reinv_timer); call->reinv_timer.id = PJ_FALSE; } break; default: call->last_code = (pjsip_status_code) e->body.tsx_state.tsx->status_code; pj_strncpy(&call->last_text, &e->body.tsx_state.tsx->status_text, sizeof(call->last_text_buf_)); break; } /* If this is an outgoing INVITE that was created because of * REFER/transfer, send NOTIFY to transferer. */ if (call->xfer_sub && e->type==PJSIP_EVENT_TSX_STATE) { int st_code = -1; pjsip_evsub_state ev_state = PJSIP_EVSUB_STATE_ACTIVE; switch (call->inv->state) { case PJSIP_INV_STATE_NULL: case PJSIP_INV_STATE_CALLING: /* Do nothing */ break; case PJSIP_INV_STATE_EARLY: case PJSIP_INV_STATE_CONNECTING: st_code = e->body.tsx_state.tsx->status_code; if (call->inv->state == PJSIP_INV_STATE_CONNECTING) ev_state = PJSIP_EVSUB_STATE_TERMINATED; else ev_state = PJSIP_EVSUB_STATE_ACTIVE; break; case PJSIP_INV_STATE_CONFIRMED: #if 0 /* We don't need this, as we've terminated the subscription in * CONNECTING state. */ /* When state is confirmed, send the final 200/OK and terminate * subscription. */ st_code = e->body.tsx_state.tsx->status_code; ev_state = PJSIP_EVSUB_STATE_TERMINATED; #endif break; case PJSIP_INV_STATE_DISCONNECTED: st_code = e->body.tsx_state.tsx->status_code; ev_state = PJSIP_EVSUB_STATE_TERMINATED; break; case PJSIP_INV_STATE_INCOMING: /* Nothing to do. Just to keep gcc from complaining about * unused enums. */ break; } if (st_code != -1) { pjsip_tx_data *tdata; pj_status_t status; status = pjsip_xfer_notify( call->xfer_sub, ev_state, st_code, NULL, &tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create NOTIFY", status); } else { status = pjsip_xfer_send_request(call->xfer_sub, tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to send NOTIFY", status); } } } } /* Release locks before calling callbacks, to avoid deadlock. */ while (PJSUA_LOCK_IS_LOCKED()) { num_locks++; PJSUA_UNLOCK(); } /* Ticket #1627: Invoke on_call_tsx_state() when call is disconnected. */ if (inv->state == PJSIP_INV_STATE_DISCONNECTED && e->type == PJSIP_EVENT_TSX_STATE && call->inv && pjsua_var.ua_cfg.cb.on_call_tsx_state) { (*pjsua_var.ua_cfg.cb.on_call_tsx_state)(call->index, e->body.tsx_state.tsx, e); } if (pjsua_var.ua_cfg.cb.on_call_state) (*pjsua_var.ua_cfg.cb.on_call_state)(call->index, e); /* Re-acquire the locks. */ for (;num_locks > 0; num_locks--) PJSUA_LOCK(); /* call->inv may be NULL now */ /* Destroy media session when invite session is disconnected. */ if (inv->state == PJSIP_INV_STATE_DISCONNECTED) { PJSUA_LOCK(); pjsua_media_channel_deinit(call->index); /* Free call */ call->inv = NULL; pj_assert(pjsua_var.call_cnt > 0); --pjsua_var.call_cnt; /* Reset call */ reset_call(call->index); pjsua_check_snd_dev_idle(); PJSUA_UNLOCK(); } pj_log_pop_indent(); } /* * This callback is called by invite session framework when UAC session * has forked. */ static void pjsua_call_on_forked( pjsip_inv_session *inv, pjsip_event *e) { PJ_UNUSED_ARG(inv); PJ_UNUSED_ARG(e); PJ_TODO(HANDLE_FORKED_DIALOG); } /* * Callback from UA layer when forked dialog response is received. */ pjsip_dialog* on_dlg_forked(pjsip_dialog *dlg, pjsip_rx_data *res) { if (dlg->uac_has_2xx && res->msg_info.cseq->method.id == PJSIP_INVITE_METHOD && pjsip_rdata_get_tsx(res) == NULL && res->msg_info.msg->line.status.code/100 == 2) { pjsip_dialog *forked_dlg; pjsip_tx_data *bye; pj_status_t status; /* Create forked dialog */ status = pjsip_dlg_fork(dlg, res, &forked_dlg); if (status != PJ_SUCCESS) return NULL; pjsip_dlg_inc_lock(forked_dlg); /* Disconnect the call */ status = pjsip_dlg_create_request(forked_dlg, pjsip_get_bye_method(), -1, &bye); if (status == PJ_SUCCESS) { status = pjsip_dlg_send_request(forked_dlg, bye, -1, NULL); } pjsip_dlg_dec_lock(forked_dlg); if (status != PJ_SUCCESS) { return NULL; } return forked_dlg; } else { return dlg; } } /* * Disconnect call upon error. */ static void call_disconnect( pjsip_inv_session *inv, int code ) { pjsip_tx_data *tdata; pj_status_t status; status = pjsip_inv_end_session(inv, code, NULL, &tdata); if (status != PJ_SUCCESS) return; #if DISABLED_FOR_TICKET_1185 pjsua_call *call; /* Add SDP in 488 status */ call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id]; if (call && call->tp && tdata->msg->type==PJSIP_RESPONSE_MSG && code==PJSIP_SC_NOT_ACCEPTABLE_HERE) { pjmedia_sdp_session *local_sdp; pjmedia_transport_info ti; pjmedia_transport_info_init(&ti); pjmedia_transport_get_info(call->med_tp, &ti); status = pjmedia_endpt_create_sdp(pjsua_var.med_endpt, tdata->pool, 1, &ti.sock_info, &local_sdp); if (status == PJ_SUCCESS) { pjsip_create_sdp_body(tdata->pool, local_sdp, &tdata->msg->body); } } #endif pjsip_inv_send_msg(inv, tdata); } /* * Callback to be called when SDP offer/answer negotiation has just completed * in the session. This function will start/update media if negotiation * has succeeded. */ static void pjsua_call_on_media_update(pjsip_inv_session *inv, pj_status_t status) { pjsua_call *call; const pjmedia_sdp_session *local_sdp; const pjmedia_sdp_session *remote_sdp; //const pj_str_t st_update = {"UPDATE", 6}; pj_log_push_indent(); call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id]; if (call->hanging_up) goto on_return; if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "SDP negotiation has failed", status); /* Clean up provisional media */ pjsua_media_prov_clean_up(call->index); /* Do not deinitialize media since this may be a re-INVITE or * UPDATE (which in this case the media should not get affected * by the failed re-INVITE/UPDATE). The media will be shutdown * when call is disconnected anyway. */ /* Stop/destroy media, if any */ /*pjsua_media_channel_deinit(call->index);*/ /* Disconnect call if we're not in the middle of initializing an * UAS dialog and if this is not a re-INVITE */ if (inv->state != PJSIP_INV_STATE_NULL && inv->state != PJSIP_INV_STATE_CONFIRMED) { call_disconnect(inv, PJSIP_SC_UNSUPPORTED_MEDIA_TYPE); } goto on_return; } /* Get local and remote SDP */ status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &local_sdp); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to retrieve currently active local SDP", status); //call_disconnect(inv, PJSIP_SC_UNSUPPORTED_MEDIA_TYPE); goto on_return; } status = pjmedia_sdp_neg_get_active_remote(call->inv->neg, &remote_sdp); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to retrieve currently active remote SDP", status); //call_disconnect(inv, PJSIP_SC_UNSUPPORTED_MEDIA_TYPE); goto on_return; } call->med_update_success = (status == PJ_SUCCESS); /* Update remote's NAT type */ if (pjsua_var.ua_cfg.nat_type_in_sdp) { update_remote_nat_type(call, remote_sdp); } /* Update media channel with the new SDP */ status = pjsua_media_channel_update(call->index, local_sdp, remote_sdp); /* If this is not the initial INVITE, don't disconnect call due to * no media after SDP negotiation. */ if (status == PJMEDIA_SDPNEG_ENOMEDIA && call->inv->state == PJSIP_INV_STATE_CONFIRMED) { status = PJ_SUCCESS; } /* Disconnect call after failure in media channel update */ if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create media session", status); call_disconnect(inv, PJSIP_SC_NOT_ACCEPTABLE_HERE); /* No need to deinitialize; media will be shutdown when call * state is disconnected anyway. */ /*pjsua_media_channel_deinit(call->index);*/ goto on_return; } /* Ticket #476: make sure only one codec is specified in the answer. */ pjsua_call_schedule_reinvite_check(call, 0); /* Call application callback, if any */ if (pjsua_var.ua_cfg.cb.on_call_media_state) pjsua_var.ua_cfg.cb.on_call_media_state(call->index); on_return: pj_log_pop_indent(); } /* Modify SDP for call hold. */ static pj_status_t modify_sdp_of_call_hold(pjsua_call *call, pj_pool_t *pool, pjmedia_sdp_session *sdp, pj_bool_t as_answerer) { unsigned mi; /* Call-hold is done by set the media direction to 'sendonly' * (PJMEDIA_DIR_ENCODING), except when current media direction is * 'inactive' (PJMEDIA_DIR_NONE). * (See RFC 3264 Section 8.4 and RFC 4317 Section 3.1) */ /* http://trac.pjsip.org/repos/ticket/880 if (call->dir != PJMEDIA_DIR_ENCODING) { */ /* https://trac.pjsip.org/repos/ticket/1142: * configuration to use c=0.0.0.0 for call hold. */ for (mi=0; mimedia_count; ++mi) { pjmedia_sdp_media *m = sdp->media[mi]; if (call->call_hold_type == PJSUA_CALL_HOLD_TYPE_RFC2543) { pjmedia_sdp_conn *conn; pjmedia_sdp_attr *attr; /* Get SDP media connection line */ conn = m->conn; if (!conn) conn = sdp->conn; /* Modify address */ conn->addr = pj_str("0.0.0.0"); /* Remove existing directions attributes */ pjmedia_sdp_media_remove_all_attr(m, "sendrecv"); pjmedia_sdp_media_remove_all_attr(m, "sendonly"); pjmedia_sdp_media_remove_all_attr(m, "recvonly"); pjmedia_sdp_media_remove_all_attr(m, "inactive"); /* Add inactive attribute */ attr = pjmedia_sdp_attr_create(pool, "inactive", NULL); pjmedia_sdp_media_add_attr(m, attr); } else { pjmedia_sdp_attr *attr; /* Remove existing directions attributes */ pjmedia_sdp_media_remove_all_attr(m, "sendrecv"); pjmedia_sdp_media_remove_all_attr(m, "sendonly"); pjmedia_sdp_media_remove_all_attr(m, "recvonly"); pjmedia_sdp_media_remove_all_attr(m, "inactive"); /* When as answerer, just simply set dir to "sendonly", note that * if the offer uses "sendonly" or "inactive", the SDP negotiator * will change our answer dir to "inactive". */ if (as_answerer || (call->media[mi].dir & PJMEDIA_DIR_ENCODING)) { /* Add sendonly attribute */ attr = pjmedia_sdp_attr_create(pool, "sendonly", NULL); pjmedia_sdp_media_add_attr(m, attr); } else { /* Add inactive attribute */ attr = pjmedia_sdp_attr_create(pool, "inactive", NULL); pjmedia_sdp_media_add_attr(m, attr); } } } return PJ_SUCCESS; } /* Create SDP for call hold. */ static pj_status_t create_sdp_of_call_hold(pjsua_call *call, pjmedia_sdp_session **p_sdp) { pj_status_t status; pj_pool_t *pool; pjmedia_sdp_session *sdp; /* Use call's provisional pool */ pool = call->inv->pool_prov; /* Create new offer */ status = pjsua_media_channel_create_sdp(call->index, pool, NULL, &sdp, NULL); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create local SDP", status); return status; } status = modify_sdp_of_call_hold(call, pool, sdp, PJ_FALSE); if (status != PJ_SUCCESS) return status; *p_sdp = sdp; return PJ_SUCCESS; } /* * Called when session received new offer. */ static void pjsua_call_on_rx_offer(pjsip_inv_session *inv, struct pjsip_inv_on_rx_offer_cb_param *param) { pjsua_call *call; pjmedia_sdp_session *answer; unsigned i; pj_status_t status; const pjmedia_sdp_session *offer = param->offer; pjsua_call_setting opt; pj_bool_t async = PJ_FALSE; call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id]; /* Supply candidate answer */ PJ_LOG(4,(THIS_FILE, "Call %d: received updated media offer", call->index)); pj_log_push_indent(); if (pjsua_call_media_is_changing(call)) { PJ_LOG(1,(THIS_FILE, "Unable to process offer" ERR_MEDIA_CHANGING)); goto on_return; } pjsua_call_cleanup_flag(&call->opt); opt = call->opt; if (pjsua_var.ua_cfg.cb.on_call_rx_reinvite && param->rdata->msg_info.msg->type == PJSIP_REQUEST_MSG && param->rdata->msg_info.msg->line.req.method.id == PJSIP_INVITE_METHOD) { pjsip_status_code code = PJSIP_SC_OK; /* If on_call_rx_reinvite() callback is implemented, * call it first. */ (*pjsua_var.ua_cfg.cb.on_call_rx_reinvite)( call->index, offer, (pjsip_rx_data *)param->rdata, NULL, &async, &code, &opt); if (async) { pjsip_tx_data *response; status = pjsip_inv_initial_answer(inv, (pjsip_rx_data *)param->rdata, 100, NULL, NULL, &response); if (status != PJ_SUCCESS) { PJ_PERROR(3, (THIS_FILE, status, "Failed to create initial answer")); goto on_return; } status = pjsip_inv_send_msg(inv, response); if (status != PJ_SUCCESS) { PJ_PERROR(3, (THIS_FILE, status, "Failed to send initial answer")); goto on_return; } PJ_LOG(4,(THIS_FILE, "App will manually answer the re-INVITE " "on call %d", call->index)); } if (code != PJSIP_SC_OK) { PJ_LOG(4,(THIS_FILE, "Rejecting re-INVITE updated media offer " "on call %d", call->index)); goto on_return; } call->opt = opt; } if (pjsua_var.ua_cfg.cb.on_call_rx_offer && !async) { pjsip_status_code code = PJSIP_SC_OK; (*pjsua_var.ua_cfg.cb.on_call_rx_offer)(call->index, offer, NULL, &code, &opt); if (code != PJSIP_SC_OK) { PJ_LOG(4,(THIS_FILE, "Rejecting updated media offer on call %d", call->index)); goto on_return; } call->opt = opt; } /* Re-init media for the new remote offer before creating SDP */ status = apply_call_setting(call, &call->opt, offer); if (status != PJ_SUCCESS) goto on_return; status = pjsua_media_channel_create_sdp(call->index, call->inv->pool_prov, offer, &answer, NULL); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create local SDP", status); goto on_return; } if (async) { call->rx_reinv_async = async; goto on_return; } /* Validate media count in the generated answer */ pj_assert(answer->media_count == offer->media_count); /* Check if offer's conn address is zero */ for (i = 0; i < answer->media_count; ++i) { pjmedia_sdp_conn *conn; conn = offer->media[i]->conn; if (!conn) conn = offer->conn; if (pj_strcmp2(&conn->addr, "0.0.0.0")==0 || pj_strcmp2(&conn->addr, "0")==0) { pjmedia_sdp_conn *a_conn = answer->media[i]->conn; /* Modify answer address */ if (a_conn) { a_conn->addr = pj_str("0.0.0.0"); } else if (answer->conn == NULL || pj_strcmp2(&answer->conn->addr, "0.0.0.0") != 0) { a_conn = PJ_POOL_ZALLOC_T(call->inv->pool_prov, pjmedia_sdp_conn); a_conn->net_type = pj_str("IN"); a_conn->addr_type = pj_str("IP4"); a_conn->addr = pj_str("0.0.0.0"); answer->media[i]->conn = a_conn; } } } /* Check if call is on-hold */ if (call->local_hold) { modify_sdp_of_call_hold(call, call->inv->pool_prov, answer, PJ_TRUE); } status = pjsip_inv_set_sdp_answer(call->inv, answer); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to set answer", status); goto on_return; } on_return: pj_log_pop_indent(); } /* * Called when receiving re-INVITE. */ static pj_status_t pjsua_call_on_rx_reinvite(pjsip_inv_session *inv, const pjmedia_sdp_session *offer, pjsip_rx_data *rdata) { pjsua_call *call; pj_bool_t async; PJ_UNUSED_ARG(offer); PJ_UNUSED_ARG(rdata); call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id]; async = call->rx_reinv_async; call->rx_reinv_async = PJ_FALSE; return (async? PJ_SUCCESS: !PJ_SUCCESS); } /* * Called to generate new offer. */ static void pjsua_call_on_create_offer(pjsip_inv_session *inv, pjmedia_sdp_session **offer) { pjsua_call *call; pj_status_t status; unsigned mi; pj_log_push_indent(); call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id]; if (pjsua_call_media_is_changing(call)) { *offer = NULL; PJ_LOG(1,(THIS_FILE, "Unable to create offer" ERR_MEDIA_CHANGING)); goto on_return; } #if RESTART_ICE_ON_REINVITE /* Ticket #1783, RFC 5245 section 12.5: * If an agent receives a mid-dialog re-INVITE that contains no offer, * it MUST restart ICE for each media stream and go through the process * of gathering new candidates. */ for (mi=0; mimed_cnt; ++mi) { pjsua_call_media *call_med = &call->media[mi]; pjmedia_transport_info tpinfo; pjmedia_ice_transport_info *ice_info; /* Check if the media is using ICE */ pjmedia_transport_info_init(&tpinfo); pjmedia_transport_get_info(call_med->tp, &tpinfo); ice_info = (pjmedia_ice_transport_info*) pjmedia_transport_info_get_spc_info( &tpinfo, PJMEDIA_TRANSPORT_TYPE_ICE); if (!ice_info) continue; /* Stop and re-init ICE stream transport. * According to RFC 5245 section 9.1.1.1, during ICE restart, * media can continue to be sent to the previously validated pair. */ pjmedia_transport_media_stop(call_med->tp); pjmedia_transport_media_create(call_med->tp, call->inv->pool_prov, (call_med->enable_rtcp_mux? PJMEDIA_TPMED_RTCP_MUX: 0), NULL, mi); PJ_LOG(4, (THIS_FILE, "Restarting ICE for media %d", mi)); } #endif pjsua_call_cleanup_flag(&call->opt); if (pjsua_var.ua_cfg.cb.on_call_tx_offer) { (*pjsua_var.ua_cfg.cb.on_call_tx_offer)(call->index, NULL, &call->opt); } /* We may need to re-initialize media before creating SDP */ if (call->med_prov_cnt == 0 || pjsua_var.ua_cfg.cb.on_call_tx_offer) { status = apply_call_setting(call, &call->opt, NULL); if (status != PJ_SUCCESS) goto on_return; } /* See if we've put call on hold. */ if (call->local_hold) { PJ_LOG(4,(THIS_FILE, "Call %d: call is on-hold locally, creating call-hold SDP ", call->index)); status = create_sdp_of_call_hold( call, offer ); } else { PJ_LOG(4,(THIS_FILE, "Call %d: asked to send a new offer", call->index)); status = pjsua_media_channel_create_sdp(call->index, call->inv->pool_prov, NULL, offer, NULL); } if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create local SDP", status); goto on_return; } on_return: pj_log_pop_indent(); } /* * Callback called by event framework when the xfer subscription state * has changed. */ static void xfer_client_on_evsub_state( pjsip_evsub *sub, pjsip_event *event) { PJ_UNUSED_ARG(event); pj_log_push_indent(); /* * 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 }; pjsua_call *call; call = (pjsua_call*) pjsip_evsub_get_mod_data(sub, pjsua_var.mod.id); /* 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 * transferred successfully. */ if (call && pjsua_var.ua_cfg.cb.on_call_transfer_status) { const pj_str_t ACCEPTED = { "Accepted", 8 }; pj_bool_t cont = PJ_FALSE; (*pjsua_var.ua_cfg.cb.on_call_transfer_status)(call->index, 200, &ACCEPTED, PJ_TRUE, &cont); } /* Yes, subscription is suppressed. * Terminate our subscription now. */ PJ_LOG(4,(THIS_FILE, "Xfer subscription suppressed, terminating " "event subcription...")); pjsip_evsub_terminate(sub, PJ_TRUE); } else { /* Notify application about call transfer progress. * Initially notify with 100/Accepted status. */ if (call && pjsua_var.ua_cfg.cb.on_call_transfer_status) { const pj_str_t ACCEPTED = { "Accepted", 8 }; pj_bool_t cont = PJ_FALSE; (*pjsua_var.ua_cfg.cb.on_call_transfer_status)(call->index, 100, &ACCEPTED, PJ_FALSE, &cont); } } } /* * On incoming NOTIFY or an error response, 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) { pjsua_call *call; 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; call = (pjsua_call*) pjsip_evsub_get_mod_data(sub, pjsua_var.mod.id); /* 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, pjsua_var.mod.id, NULL); PJ_LOG(4,(THIS_FILE, "Xfer client subscription terminated")); } if (!call || !event || !pjsua_var.ua_cfg.cb.on_call_transfer_status) { /* Application is not interested with call progress status */ goto on_return; } 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; msg = rdata->msg_info.msg; /* This better be a NOTIFY request */ if (pjsip_method_cmp(&msg->line.req.method, pjsip_get_notify_method()) == 0) { /* Check if there's body */ body = msg->body; if (!body) { PJ_LOG(2, (THIS_FILE, "Warning: received NOTIFY without message " "body")); goto on_return; } /* Check for appropriate content */ if (pj_stricmp2(&body->content_type.type, "message") != 0 || pj_stricmp2(&body->content_type.subtype, "sipfrag") != 0) { PJ_LOG(2, (THIS_FILE, "Warning: received NOTIFY with non " "message/sipfrag content")); goto on_return; } /* Try to parse the content */ status = pjsip_parse_status_line((char*)body->data, body->len, &status_line); if (status != PJ_SUCCESS) { PJ_LOG(2, (THIS_FILE, "Warning: received NOTIFY with invalid " "message/sipfrag content")); goto on_return; } } else { status_line.code = msg->line.status.code; status_line.reason = msg->line.status.reason; } } else { 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; (*pjsua_var.ua_cfg.cb.on_call_transfer_status)(call->index, status_line.code, &status_line.reason, is_last, &cont); if (!cont) { pjsip_evsub_set_mod_data(sub, pjsua_var.mod.id, NULL); } /* If the call transfer has completed but the subscription is * not terminated, terminate it now. */ if (status_line.code/100 == 2 && !is_last) { pjsip_tx_data *tdata; status = pjsip_evsub_initiate(sub, pjsip_get_subscribe_method(), 0, &tdata); if (status == PJ_SUCCESS) status = pjsip_evsub_send_request(sub, tdata); } } on_return: pj_log_pop_indent(); } /* * Callback called by event framework when the xfer subscription state * has changed. */ static void xfer_server_on_evsub_state( pjsip_evsub *sub, pjsip_event *event) { PJ_UNUSED_ARG(event); pj_log_push_indent(); /* * When subscription is terminated, clear the xfer_sub member of * the inv_data. */ if (pjsip_evsub_get_state(sub) == PJSIP_EVSUB_STATE_TERMINATED) { pjsua_call *call; call = (pjsua_call*) pjsip_evsub_get_mod_data(sub, pjsua_var.mod.id); if (!call) goto on_return; pjsip_evsub_set_mod_data(sub, pjsua_var.mod.id, NULL); call->xfer_sub = NULL; PJ_LOG(4,(THIS_FILE, "Xfer server subscription terminated")); } on_return: pj_log_pop_indent(); } /* * Follow transfer (REFER) request. */ static void on_call_transferred( pjsip_inv_session *inv, pjsip_rx_data *rdata ) { pj_status_t status; pjsip_tx_data *tdata; pjsua_call *existing_call; int new_call; const pj_str_t str_refer_to = { "Refer-To", 8}; const pj_str_t str_refer_sub = { "Refer-Sub", 9 }; const pj_str_t str_ref_by = { "Referred-By", 11 }; pjsip_generic_string_hdr *refer_to; pjsip_generic_string_hdr *refer_sub; pjsip_hdr *ref_by_hdr; pj_bool_t no_refer_sub = PJ_FALSE; char *uri; pjsua_msg_data msg_data; pj_str_t tmp; pjsip_status_code code; pjsip_evsub *sub; pjsua_call_setting call_opt; pj_log_push_indent(); existing_call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id]; /* Find the Refer-To header */ refer_to = (pjsip_generic_string_hdr*) pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_refer_to, NULL); if (refer_to == NULL) { /* Invalid Request. * No Refer-To header! */ PJ_LOG(4,(THIS_FILE, "Received REFER without Refer-To header!")); pjsip_dlg_respond( inv->dlg, rdata, 400, NULL, NULL, NULL); goto on_return; } /* Find optional Refer-Sub header */ refer_sub = (pjsip_generic_string_hdr*) pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_refer_sub, NULL); if (refer_sub) { if (pj_strnicmp2(&refer_sub->hvalue, "true", 4)!=0) no_refer_sub = PJ_TRUE; } /* Find optional Referred-By header (to be copied onto outgoing INVITE * request. */ ref_by_hdr = (pjsip_hdr*) pjsip_msg_find_hdr_by_name(rdata->msg_info.msg, &str_ref_by, NULL); /* Notify callback */ code = PJSIP_SC_ACCEPTED; if (pjsua_var.ua_cfg.cb.on_call_transfer_request) { (*pjsua_var.ua_cfg.cb.on_call_transfer_request)(existing_call->index, &refer_to->hvalue, &code); } pjsua_call_cleanup_flag(&existing_call->opt); call_opt = existing_call->opt; if (pjsua_var.ua_cfg.cb.on_call_transfer_request2) { (*pjsua_var.ua_cfg.cb.on_call_transfer_request2)(existing_call->index, &refer_to->hvalue, &code, &call_opt); } if (code < 200) code = PJSIP_SC_ACCEPTED; if (code >= 300) { /* Application rejects call transfer request */ pjsip_dlg_respond( inv->dlg, rdata, code, NULL, NULL, NULL); goto on_return; } PJ_LOG(3,(THIS_FILE, "Call to %.*s is being transferred to %.*s", (int)inv->dlg->remote.info_str.slen, inv->dlg->remote.info_str.ptr, (int)refer_to->hvalue.slen, refer_to->hvalue.ptr)); if (no_refer_sub) { /* * Always answer with 2xx. */ pjsip_tx_data *tdata2; const pj_str_t str_false = { "false", 5}; pjsip_hdr *hdr; status = pjsip_dlg_create_response(inv->dlg, rdata, code, NULL, &tdata2); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create 2xx response to REFER", status); goto on_return; } /* Add Refer-Sub header */ hdr = (pjsip_hdr*) pjsip_generic_string_hdr_create(tdata2->pool, &str_refer_sub, &str_false); pjsip_msg_add_hdr(tdata2->msg, hdr); /* Send answer */ status = pjsip_dlg_send_response(inv->dlg, pjsip_rdata_get_tsx(rdata), tdata2); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create 2xx response to REFER", status); goto on_return; } /* Don't have subscription */ sub = NULL; } else { struct pjsip_evsub_user xfer_cb; pjsip_hdr hdr_list; /* Init callback */ pj_bzero(&xfer_cb, sizeof(xfer_cb)); xfer_cb.on_evsub_state = &xfer_server_on_evsub_state; /* Init additional header list to be sent with REFER response */ pj_list_init(&hdr_list); /* Create transferee event subscription */ status = pjsip_xfer_create_uas( inv->dlg, &xfer_cb, rdata, &sub); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create xfer uas", status); pjsip_dlg_respond( inv->dlg, rdata, 500, NULL, NULL, NULL); goto on_return; } /* If there's Refer-Sub header and the value is "true", send back * Refer-Sub in the response with value "true" too. */ if (refer_sub) { const pj_str_t str_true = { "true", 4 }; pjsip_hdr *hdr; hdr = (pjsip_hdr*) pjsip_generic_string_hdr_create(inv->dlg->pool, &str_refer_sub, &str_true); pj_list_push_back(&hdr_list, hdr); } /* Accept the REFER request, send 2xx. */ pjsip_xfer_accept(sub, rdata, code, &hdr_list); /* Create initial NOTIFY request */ status = pjsip_xfer_notify( sub, PJSIP_EVSUB_STATE_ACTIVE, 100, NULL, &tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create NOTIFY to REFER", status); goto on_return; } /* Send initial NOTIFY request */ status = pjsip_xfer_send_request( sub, tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to send NOTIFY to REFER", status); goto on_return; } } /* We're cheating here. * We need to get a null terminated string from a pj_str_t. * So grab the pointer from the hvalue and NULL terminate it, knowing * that the NULL position will be occupied by a newline. */ uri = refer_to->hvalue.ptr; uri[refer_to->hvalue.slen] = '\0'; /* Init msg_data */ pjsua_msg_data_init(&msg_data); /* If Referred-By header is present in the REFER request, copy this * to the outgoing INVITE request. */ if (ref_by_hdr != NULL) { pjsip_hdr *dup = (pjsip_hdr*) pjsip_hdr_clone(rdata->tp_info.pool, ref_by_hdr); pj_list_push_back(&msg_data.hdr_list, dup); } /* Now make the outgoing call. */ tmp = pj_str(uri); status = pjsua_call_make_call(existing_call->acc_id, &tmp, &call_opt, existing_call->user_data, &msg_data, &new_call); if (status != PJ_SUCCESS) { /* Notify xferer about the error (if we have subscription) */ if (sub) { status = pjsip_xfer_notify(sub, PJSIP_EVSUB_STATE_TERMINATED, 500, NULL, &tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to create NOTIFY to REFER", status); goto on_return; } status = pjsip_xfer_send_request(sub, tdata); if (status != PJ_SUCCESS) { pjsua_perror(THIS_FILE, "Unable to send NOTIFY to REFER", status); goto on_return; } } goto on_return; } if (sub) { /* Put the server subscription in inv_data. * Subsequent state changed in pjsua_inv_on_state_changed() will be * reported back to the server subscription. */ pjsua_var.calls[new_call].xfer_sub = sub; /* Put the invite_data in the subscription. */ pjsip_evsub_set_mod_data(sub, pjsua_var.mod.id, &pjsua_var.calls[new_call]); } on_return: pj_log_pop_indent(); } /* * This callback is called when transaction state has changed in INVITE * session. We use this to trap: * - incoming REFER request. * - incoming MESSAGE request. */ static void pjsua_call_on_tsx_state_changed(pjsip_inv_session *inv, pjsip_transaction *tsx, pjsip_event *e) { pjsua_call *call; pj_log_push_indent(); call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id]; if (call == NULL) goto on_return; if (call->inv == NULL) { /* Call has been disconnected. */ goto on_return; } /* https://trac.pjsip.org/repos/ticket/1452: * If a request is retried due to 401/407 challenge, don't process the * transaction first but wait until we've retried it. */ if (tsx->role == PJSIP_ROLE_UAC && (tsx->status_code==401 || tsx->status_code==407) && tsx->last_tx && tsx->last_tx->auth_retry) { goto on_return; } /* Notify application callback first */ if (pjsua_var.ua_cfg.cb.on_call_tsx_state) { (*pjsua_var.ua_cfg.cb.on_call_tsx_state)(call->index, tsx, e); } if (tsx->role==PJSIP_ROLE_UAS && tsx->state==PJSIP_TSX_STATE_TRYING && pjsip_method_cmp(&tsx->method, pjsip_get_refer_method())==0) { /* * Incoming REFER request. */ on_call_transferred(call->inv, e->body.tsx_state.src.rdata); } else if (tsx->role==PJSIP_ROLE_UAS && tsx->state==PJSIP_TSX_STATE_TRYING && pjsip_method_cmp(&tsx->method, &pjsip_message_method)==0) { /* * Incoming MESSAGE request! */ pjsip_rx_data *rdata; pjsip_accept_hdr *accept_hdr; rdata = e->body.tsx_state.src.rdata; /* Request MUST have message body, with Content-Type equal to * "text/plain". */ if (pjsua_im_accept_pager(rdata, &accept_hdr) == PJ_FALSE) { pjsip_hdr hdr_list; pj_list_init(&hdr_list); pj_list_push_back(&hdr_list, accept_hdr); pjsip_dlg_respond( inv->dlg, rdata, PJSIP_SC_NOT_ACCEPTABLE_HERE, NULL, &hdr_list, NULL ); goto on_return; } /* Respond with 200 first, so that remote doesn't retransmit in case * the UI takes too long to process the message. */ pjsip_dlg_respond( inv->dlg, rdata, 200, NULL, NULL, NULL); /* Process MESSAGE request */ pjsua_im_process_pager(call->index, &inv->dlg->remote.info_str, &inv->dlg->local.info_str, rdata); } else if (tsx->role == PJSIP_ROLE_UAC && pjsip_method_cmp(&tsx->method, &pjsip_message_method)==0) { /* Handle outgoing pager status */ if (tsx->status_code >= 200) { pjsua_im_data *im_data; im_data = (pjsua_im_data*) tsx->mod_data[pjsua_var.mod.id]; /* im_data can be NULL if this is typing indication */ if (im_data && pjsua_var.ua_cfg.cb.on_pager_status) { pjsua_var.ua_cfg.cb.on_pager_status(im_data->call_id, &im_data->to, &im_data->body, im_data->user_data, (pjsip_status_code) tsx->status_code, &tsx->status_text); } } } else if (tsx->role == PJSIP_ROLE_UAC && pjsip_method_cmp(&tsx->method, pjsip_get_invite_method())==0 && tsx->state >= PJSIP_TSX_STATE_COMPLETED && e->body.tsx_state.prev_state < PJSIP_TSX_STATE_COMPLETED && (!PJSIP_IS_STATUS_IN_CLASS(tsx->status_code, 300) && tsx->status_code!=401 && tsx->status_code!=407 && tsx->status_code!=422)) { if (tsx->status_code/100 == 2) { /* If we have sent CANCEL and the original INVITE returns a 2xx, * we then send BYE. */ if (call->hanging_up) { PJ_LOG(3,(THIS_FILE, "Unsuccessful in cancelling the original " "INVITE for call %d due to %d response, sending BYE " "instead", call->index, tsx->status_code)); call_disconnect(call->inv, PJSIP_SC_OK); } } else { /* Monitor the status of call hold/unhold request */ if (tsx->last_tx == (pjsip_tx_data*)call->hold_msg) { /* Outgoing call hold failed */ call->local_hold = PJ_FALSE; PJ_LOG(3,(THIS_FILE, "Error putting call %d on hold " "(reason=%d)", call->index, tsx->status_code)); } else if (call->opt.flag & PJSUA_CALL_UNHOLD) { /* Call unhold failed */ call->local_hold = PJ_TRUE; PJ_LOG(3,(THIS_FILE, "Error releasing hold on call %d " "(reason=%d)", call->index, tsx->status_code)); } } if (tsx->last_tx == (pjsip_tx_data*)call->hold_msg) { call->hold_msg = NULL; } if (tsx->last_tx->msg->body && (tsx->status_code/100 != 2 || !call->med_update_success)) { /* Either we get non-2xx or media update failed, * clean up provisional media. */ pjsua_media_prov_clean_up(call->index); } } else if (tsx->role == PJSIP_ROLE_UAC && pjsip_method_cmp(&tsx->method, &pjsip_update_method)==0 && tsx->state >= PJSIP_TSX_STATE_COMPLETED && e->body.tsx_state.prev_state < PJSIP_TSX_STATE_COMPLETED && (!PJSIP_IS_STATUS_IN_CLASS(tsx->status_code, 300) && tsx->status_code!=401 && tsx->status_code!=407 && tsx->status_code!=422)) { if (tsx->last_tx->msg->body && (tsx->status_code/100 != 2 || !call->med_update_success)) { /* Either we get non-2xx or media update failed, * clean up provisional media. */ pjsua_media_prov_clean_up(call->index); } } else if (tsx->role==PJSIP_ROLE_UAS && tsx->state==PJSIP_TSX_STATE_TRYING && pjsip_method_cmp(&tsx->method, &pjsip_info_method)==0) { /* * Incoming INFO request for media control. */ const pj_str_t STR_APPLICATION = { "application", 11}; const pj_str_t STR_MEDIA_CONTROL_XML = { "media_control+xml", 17 }; /* * Incoming INFO request for DTMF. */ const pj_str_t STR_DTMF_RELAY = { "dtmf-relay", 10 }; pjsip_rx_data *rdata = e->body.tsx_state.src.rdata; pjsip_msg_body *body = rdata->msg_info.msg->body; if (body && body->len && pj_stricmp(&body->content_type.type, &STR_APPLICATION)==0 && pj_stricmp(&body->content_type.subtype, &STR_MEDIA_CONTROL_XML)==0) { pjsip_tx_data *tdata; pj_str_t control_st; pj_status_t status; /* Apply and answer the INFO request */ pj_strset(&control_st, (char*)body->data, body->len); status = pjsua_media_apply_xml_control(call->index, &control_st); if (status == PJ_SUCCESS) { status = pjsip_endpt_create_response(tsx->endpt, rdata, 200, NULL, &tdata); if (status == PJ_SUCCESS) status = pjsip_tsx_send_msg(tsx, tdata); } else { status = pjsip_endpt_create_response(tsx->endpt, rdata, 400, NULL, &tdata); if (status == PJ_SUCCESS) status = pjsip_tsx_send_msg(tsx, tdata); } } else if (body && body->len && pj_stricmp(&body->content_type.type, &STR_APPLICATION)==0 && pj_stricmp(&body->content_type.subtype, &STR_DTMF_RELAY)==0) { pjsip_tx_data *tdata; pj_status_t status; pj_bool_t is_handled = PJ_FALSE; if (pjsua_var.ua_cfg.cb.on_dtmf_digit2) { pjsua_dtmf_info info; pj_str_t delim, token, input; pj_ssize_t found_idx; delim = pj_str("\r\n"); input = pj_str(rdata->msg_info.msg->body->data); found_idx = pj_strtok(&input, &delim, &token, 0); if (found_idx != input.slen) { /* Get signal/digit */ const pj_str_t STR_SIGNAL = { "Signal=", 7 }; const pj_str_t STR_DURATION = { "Duration=", 9 }; char *val; val = pj_strstr(&input, &STR_SIGNAL); if (val) { info.digit = *(val+STR_SIGNAL.slen); is_handled = PJ_TRUE; /* Get duration */ input.ptr += token.slen + 2; input.slen -= (token.slen + 2); val = pj_strstr(&input, &STR_DURATION); if (val) { pj_str_t val_str; val_str.ptr = val + STR_DURATION.slen; val_str.slen = input.slen - STR_DURATION.slen; info.duration = pj_strtoul(&val_str); } info.method = PJSUA_DTMF_METHOD_SIP_INFO; (*pjsua_var.ua_cfg.cb.on_dtmf_digit2)(call->index, &info); status = pjsip_endpt_create_response(tsx->endpt, rdata, 200, NULL, &tdata); if (status == PJ_SUCCESS) status = pjsip_tsx_send_msg(tsx, tdata); } } } if (!is_handled) { status = pjsip_endpt_create_response(tsx->endpt, rdata, 400, NULL, &tdata); if (status == PJ_SUCCESS) status = pjsip_tsx_send_msg(tsx, tdata); } } } else if (tsx->role == PJSIP_ROLE_UAC && pjsip_method_cmp(&tsx->method, &pjsip_info_method)==0 && (tsx->state == PJSIP_TSX_STATE_COMPLETED || (tsx->state == PJSIP_TSX_STATE_TERMINATED && e->body.tsx_state.prev_state != PJSIP_TSX_STATE_COMPLETED))) { const pj_str_t STR_APPLICATION = { "application", 11}; const pj_str_t STR_DTMF_RELAY = { "dtmf-relay", 10 }; pjsip_msg_body *body = NULL; pj_bool_t dtmf_info = PJ_FALSE; if (e->body.tsx_state.type == PJSIP_EVENT_TX_MSG) body = e->body.tsx_state.src.tdata->msg->body; else body = e->body.tsx_state.tsx->last_tx->msg->body; /* Check DTMF content in the INFO message */ if (body && body->len && pj_stricmp(&body->content_type.type, &STR_APPLICATION)==0 && pj_stricmp(&body->content_type.subtype, &STR_DTMF_RELAY)==0) { dtmf_info = PJ_TRUE; } if (dtmf_info && (tsx->state == PJSIP_TSX_STATE_COMPLETED || (tsx->state == PJSIP_TSX_STATE_TERMINATED && e->body.tsx_state.prev_state != PJSIP_TSX_STATE_COMPLETED))) { /* Status of outgoing INFO request */ if (tsx->status_code >= 200 && tsx->status_code < 300) { PJ_LOG(4,(THIS_FILE, "Call %d: DTMF sent successfully with INFO", call->index)); } else if (tsx->status_code >= 300) { PJ_LOG(4,(THIS_FILE, "Call %d: Failed to send DTMF with INFO: %d/%.*s", call->index, tsx->status_code, (int)tsx->status_text.slen, tsx->status_text.ptr)); } } } on_return: pj_log_pop_indent(); } /* Redirection handler */ static pjsip_redirect_op pjsua_call_on_redirected(pjsip_inv_session *inv, const pjsip_uri *target, const pjsip_event *e) { pjsua_call *call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id]; pjsip_redirect_op op; pj_log_push_indent(); if (pjsua_var.ua_cfg.cb.on_call_redirected) { op = (*pjsua_var.ua_cfg.cb.on_call_redirected)(call->index, target, e); } else { PJ_LOG(4,(THIS_FILE, "Unhandled redirection for call %d " "(callback not implemented by application). Disconnecting " "call.", call->index)); op = PJSIP_REDIRECT_STOP; } pj_log_pop_indent(); return op; }