diff options
Diffstat (limited to 'pjsip/src/pjsua-lib/pjsua_call.c')
| -rw-r--r-- | pjsip/src/pjsua-lib/pjsua_call.c | 4397 |
1 files changed, 4397 insertions, 0 deletions
diff --git a/pjsip/src/pjsua-lib/pjsua_call.c b/pjsip/src/pjsua-lib/pjsua_call.c new file mode 100644 index 0000000..6f14709 --- /dev/null +++ b/pjsip/src/pjsua-lib/pjsua_call.c @@ -0,0 +1,4397 @@ +/* $Id: pjsua_call.c 4176 2012-06-23 03:06:52Z nanang $ */ +/* + * Copyright (C) 2008-2011 Teluu Inc. (http://www.teluu.com) + * Copyright (C) 2003-2008 Benny Prijono <benny@prijono.org> + * + * 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 <pjsua-lib/pjsua.h> +#include <pjsua-lib/pjsua_internal.h> + + +#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 + + +/* + * The INFO method. + */ +const pjsip_method pjsip_info_method = +{ + PJSIP_OTHER_METHOD, + { "INFO", 4 } +}; + + +/* 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, + const pjmedia_sdp_session *offer); + +/* + * 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); + +/* + * Reset call descriptor. + */ +static void reset_call(pjsua_call_id id) +{ + pjsua_call *call = &pjsua_var.calls[id]; + unsigned i; + + pj_bzero(call, sizeof(*call)); + call->index = id; + call->last_text.ptr = call->last_text_buf_; + for (i=0; i<PJ_ARRAY_SIZE(call->media); ++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->call = call; + call_med->idx = i; + call_med->tp_auto_del = PJ_TRUE; + } + pjsua_call_setting_default(&call->opt); +} + + +/* + * 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<PJ_ARRAY_SIZE(pjsua_var.calls); ++i) + reset_call(i); + + /* Copy config */ + pjsua_config_dup(pjsua_var.pool, &pjsua_var.ua_cfg, cfg); + + /* Verify settings */ + if (pjsua_var.ua_cfg.max_calls >= 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<pjsua_var.ua_cfg.outbound_proxy_cnt; ++i) { + status = normalize_route_uri(pjsua_var.pool, + &pjsua_var.ua_cfg.outbound_proxy[i]); + if (status != PJ_SUCCESS) + return status; + } + + /* Initialize invite session callback. */ + pj_bzero(&inv_cb, sizeof(inv_cb)); + inv_cb.on_state_changed = &pjsua_call_on_state_changed; + inv_cb.on_new_session = &pjsua_call_on_forked; + inv_cb.on_media_update = &pjsua_call_on_media_update; + inv_cb.on_rx_offer = &pjsua_call_on_rx_offer; + inv_cb.on_create_offer = &pjsua_call_on_create_offer; + inv_cb.on_tsx_state_changed = &pjsua_call_on_tsx_state_changed; + inv_cb.on_redirected = &pjsua_call_on_redirected; + + /* Initialize invite session module: */ + status = pjsip_inv_usage_init(pjsua_var.endpt, &inv_cb); + PJ_ASSERT_RETURN(status == PJ_SUCCESS, status); + + /* Add "norefersub" in Supported header */ + pjsip_endpt_add_capability(pjsua_var.endpt, NULL, PJSIP_H_SUPPORTED, + NULL, 1, &str_norefersub); + + return status; +} + + +/* + * Start call subsystem. + */ +pj_status_t pjsua_call_subsys_start(void) +{ + /* Nothing to do */ + return PJ_SUCCESS; +} + + +/* + * Get maximum number of calls configured in pjsua. + */ +PJ_DEF(unsigned) pjsua_call_get_max_count(void) +{ + return pjsua_var.ua_cfg.max_calls; +} + + +/* + * Get number of currently active calls. + */ +PJ_DEF(unsigned) pjsua_call_get_count(void) +{ + return pjsua_var.call_cnt; +} + + +/* + * Enum calls. + */ +PJ_DEF(pj_status_t) pjsua_enum_calls( pjsua_call_id ids[], + unsigned *count) +{ + unsigned i, c; + + PJ_ASSERT_RETURN(ids && *count, PJ_EINVAL); + + PJSUA_LOCK(); + + for (i=0, c=0; c<*count && i<pjsua_var.ua_cfg.max_calls; ++i) { + if (!pjsua_var.calls[i].inv) + continue; + ids[c] = i; + ++c; + } + + *count = c; + + PJSUA_UNLOCK(); + + return PJ_SUCCESS; +} + + +/* Allocate one call id */ +static pjsua_call_id alloc_call_id(void) +{ + pjsua_call_id cid; + +#if 1 + /* New algorithm: round-robin */ + if (pjsua_var.next_call_id >= (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; + 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_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; + int 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. */ + if (call->async_call.med_ch_deinit) + goto on_error; + + /* Create offer */ + 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) + 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) { + pjsua_perror(THIS_FILE, "Unable to send initial INVITE request", + status); + + /* Upon failure to send first request, the invite + * session would have been cleared. + */ + inv = NULL; + goto on_error; + } + + /* Done. */ + + pjsip_dlg_dec_lock(dlg); + PJSUA_UNLOCK(); + + return PJ_SUCCESS; + +on_error: + if (inv == NULL && call_id != -1 && pjsua_var.ua_cfg.cb.on_call_state) + (*pjsua_var.ua_cfg.cb.on_call_state)(call_id, NULL); + + 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) { + reset_call(call_id); + pjsua_media_channel_deinit(call_id); + } + + PJSUA_UNLOCK(); + return status; +} + + +/* + * 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 +} + +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) + return PJ_SUCCESS; + +#if !PJMEDIA_HAS_VIDEO + pj_assert(opt->vid_cnt == 0); +#endif + + /* If call is established, reinit media channel */ + if (call->inv && call->inv->state == PJSIP_INV_STATE_CONFIRMED) { + pjsua_call_setting old_opt; + pj_status_t status; + + old_opt = call->opt; + call->opt = *opt; + + /* Reinit media channel when media count is changed or we are the + * answerer (as remote offer may 'extremely' modify the existing + * media session, e.g: media type order). + */ + if (rem_sdp || + opt->aud_cnt!=old_opt.aud_cnt || opt->vid_cnt!=old_opt.vid_cnt) + { + pjsip_role_e role = rem_sdp? PJSIP_ROLE_UAS : PJSIP_ROLE_UAC; + 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; + } + } + } else { + call->opt = *opt; + } + + return PJ_SUCCESS; +} + +/* + * 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(); + + /* 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) + { + status = pjsua_set_snd_dev(pjsua_var.cap_dev, pjsua_var.play_dev); + if (status != PJ_SUCCESS) + goto on_error; + } + + 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; + } + + call = &pjsua_var.calls[call_id]; + + /* Associate session with account */ + call->acc_id = acc_id; + call->call_hold_type = acc->cfg.call_hold_type; + + /* 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 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, 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); + + 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); + + /* 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); + + /* 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) { + reset_call(call_id); + pjsua_media_channel_deinit(call_id); + } + + 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; + pj_status_t status; + + /* 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); + + 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) + 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; + + 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; + } +} + + +/* Incoming call callback when media transport creation is completed. */ +static pj_status_t +on_incoming_call_med_tp_complete(pjsua_call_id call_id, + const pjsua_med_tp_state_info *info) +{ + pjsua_call *call = &pjsua_var.calls[call_id]; + const pjmedia_sdp_session *offer=NULL; + pjmedia_sdp_session *answer; + pjsip_tx_data *response = NULL; + unsigned options = 0; + 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; + } + + /* Get remote SDP offer (if any). */ + if (call->inv->neg) + pjmedia_sdp_neg_get_neg_remote(call->inv->neg, &offer); + + status = pjsua_media_channel_create_sdp(call_id, + call->async_call.dlg->pool, + offer, &answer, &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, answer); + if (status != PJ_SUCCESS) { + pjsua_perror(THIS_FILE, "Error setting local SDP", status); + sip_err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE; + goto on_return; + } + + /* Verify that we can handle the request. */ + status = pjsip_inv_verify_request3(NULL, + 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. + */ + sip_err_code = PJSIP_ERRNO_TO_SIP_STATUS(status); + 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_; + + 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) { + 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); + } + } + + PJSUA_UNLOCK(); + return status; +} + + +/** + * 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; + int call_id = -1; + int sip_err_code; + pjmedia_sdp_session *offer=NULL; + 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]; + + /* 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; + + /* 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; + } + } + + /* + * 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); + 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) + status = PJSIP_ERRNO_FROM_SIP_STATUS(PJSIP_SC_NOT_ACCEPTABLE); + + if (status != PJ_SUCCESS) { + const pj_str_t reason = pj_str("Bad SDP"); + pjsip_hdr hdr_list; + 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->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.media_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( 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); + } + + /* 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); + + /* 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) { + 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_SUCCESS) { + status = on_incoming_call_med_tp_complete(call_id, NULL); + if (status != PJ_SUCCESS) { + sip_err_code = PJSIP_SC_NOT_ACCEPTABLE; + /* 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_respond(dlg, rdata, sip_err_code, NULL, NULL, NULL); + pjsip_inv_terminate(call->inv, sip_err_code, PJ_FALSE); + call->inv = NULL; + goto on_return; + } + } else if (status != PJ_EPENDING) { + pjsua_perror(THIS_FILE, "Error initializing media channel", status); + pjsip_dlg_respond(dlg, rdata, sip_err_code, NULL, NULL, NULL); + pjsip_inv_terminate(call->inv, sip_err_code, PJ_FALSE); + call->inv = 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; + + 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; + 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; + goto on_return; + } + } + + /* Only do this after sending 100/Trying (really! see the long comment + * above) + */ + 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); + } else { + pjsua_call_hangup(call_id, PJSIP_SC_TEMPORARILY_UNAVAILABLE, + NULL, NULL); + } + } + + + /* This INVITE request has been handled. */ +on_return: + 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; + + 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 == 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(call->inv->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 = call->inv->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)); + + /* 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; + } 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; + + 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_; + + 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. + */ + if (call->med_ch_cb) { + 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) { + 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; +} + + +/* + * 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; + + 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); + 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); + goto on_return; + } + + /* Stop lock codec timer, if it is active */ + if (call->lock_codec.reinv_timer.id) { + pjsip_endpt_cancel_timer(pjsua_var.endpt, + &call->lock_codec.reinv_timer); + call->lock_codec.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) +{ + pjmedia_sdp_session *sdp; + 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, "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; + } + + status = create_sdp_of_call_hold(call, &sdp); + if (status != PJ_SUCCESS) + goto on_return; + + /* Create re-INVITE with new offer */ + status = pjsip_inv_reinvite( call->inv, NULL, 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; + +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, NULL, 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; + 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 (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 { + status = pjsua_media_channel_create_sdp(call->index, + call->inv->pool_prov, + NULL, &sdp, NULL); + call->local_hold = PJ_FALSE; + } + 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)) + { + new_contact = &pjsua_var.acc[call->acc_id].contact; + } + + /* 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 */ + status = pjsip_inv_send_msg( call->inv, tdata); + 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, NULL, 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; + 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; + + 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 */ + 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)) + { + new_contact = &pjsua_var.acc[call->acc_id].contact; + } + + /* 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 */ + status = pjsip_inv_send_msg( call->inv, tdata); + if (status != PJ_SUCCESS) { + pjsua_perror(THIS_FILE, "Unable to send UPDATE request", status); + goto on_return; + } + + call->local_hold = PJ_FALSE; + +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, "Transfering 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; + pjsip_uri *uri; + pj_status_t status; + + + 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, "Transfering 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; + + + /* 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&"), + (int)dest_dlg->call_id->id.slen, + dest_dlg->call_id->id.ptr, + (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 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; i<pjsua_var.ua_cfg.max_calls; ++i) { + if (pjsua_var.calls[i].inv) + pjsua_call_hangup(i, 0, NULL, NULL); + } + + //PJSUA_UNLOCK(); + pj_log_pop_indent(); +} + + +/* Proto */ +static pj_status_t perform_lock_codec(pjsua_call *call); + +/* Timer callback to send re-INVITE or UPDATE to lock codec */ +static void reinv_timer_cb(pj_timer_heap_t *th, + pj_timer_entry *entry) +{ + pjsua_call_id call_id = (pjsua_call_id)(pj_size_t)entry->user_data; + pjsip_dialog *dlg; + pjsua_call *call; + pj_status_t status; + + PJ_UNUSED_ARG(th); + + pjsua_var.calls[call_id].lock_codec.reinv_timer.id = PJ_FALSE; + + status = acquire_call("reinv_timer_cb()", call_id, &call, &dlg); + if (status != PJ_SUCCESS) + return; + + status = perform_lock_codec(call); + + pjsip_dlg_dec_lock(dlg); +} + + +/* 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; +} + + +/* Send re-INVITE or UPDATE with new SDP offer to select only one codec + * out of several codecs presented by callee in his answer. + */ +static pj_status_t perform_lock_codec(pjsua_call *call) +{ + const pj_str_t STR_UPDATE = {"UPDATE", 6}; + const pjmedia_sdp_session *local_sdp = NULL, *new_sdp; + unsigned i; + pj_bool_t rem_can_update; + pj_bool_t need_lock_codec = PJ_FALSE; + pjsip_tx_data *tdata; + pj_status_t status; + + PJ_ASSERT_RETURN(call->lock_codec.reinv_timer.id==PJ_FALSE, + PJ_EINVALIDOP); + + /* Verify if another SDP negotiation is in progress, e.g: session timer + * or another re-INVITE. + */ + if (call->inv==NULL || call->inv->neg==NULL || + pjmedia_sdp_neg_get_state(call->inv->neg)!=PJMEDIA_SDP_NEG_STATE_DONE) + { + return PJMEDIA_SDPNEG_EINSTATE; + } + + /* Don't do this if call is disconnecting! */ + if (call->inv->state > PJSIP_INV_STATE_CONFIRMED || + call->inv->cause >= 200) + { + return PJ_EINVALIDOP; + } + + /* Verify if another SDP negotiation has been completed by comparing + * the SDP version. + */ + status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &local_sdp); + if (status != PJ_SUCCESS) + return status; + if (local_sdp->origin.version > call->lock_codec.sdp_ver) + return PJMEDIA_SDP_EINVER; + + PJ_LOG(3, (THIS_FILE, "Updating media session to use only one codec..")); + + /* Update the new offer so it contains only a codec. Note that formats + * order in the offer should have been matched to the answer, so we can + * just directly update the offer without looking-up the answer. + */ + new_sdp = pjmedia_sdp_session_clone(call->inv->pool_prov, local_sdp); + + for (i = 0; i < call->med_cnt; ++i) { + unsigned j = 0, codec_cnt = 0; + const pjmedia_sdp_media *ref_m; + pjmedia_sdp_media *m; + 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; + } + + ref_m = local_sdp->media[i]; + m = new_sdp->media[i]; + + /* Verify that media must be active. */ + pj_assert(ref_m->desc.port); + + while (j < m->desc.fmt_count) { + pjmedia_sdp_attr *a; + pj_str_t *fmt = &m->desc.fmt[j]; + + if (is_non_av_fmt(m, fmt) || (++codec_cnt == 1)) { + ++j; + continue; + } + + /* Remove format */ + a = pjmedia_sdp_attr_find2(m->attr_count, m->attr, "rtpmap", fmt); + if (a) pjmedia_sdp_attr_remove(&m->attr_count, m->attr, a); + a = pjmedia_sdp_attr_find2(m->attr_count, m->attr, "fmtp", fmt); + if (a) pjmedia_sdp_attr_remove(&m->attr_count, m->attr, a); + pj_array_erase(m->desc.fmt, sizeof(m->desc.fmt[0]), + m->desc.fmt_count, j); + --m->desc.fmt_count; + } + + need_lock_codec |= (ref_m->desc.fmt_count > m->desc.fmt_count); + } + + /* Last check if SDP trully needs to be updated. It is possible that OA + * negotiations have completed and SDP has changed but we didn't + * increase the SDP version (should not happen!). + */ + if (!need_lock_codec) + return PJ_SUCCESS; + + /* Send UPDATE or re-INVITE */ + rem_can_update = pjsip_dlg_remote_has_cap(call->inv->dlg, + PJSIP_H_ALLOW, + NULL, &STR_UPDATE) == + PJSIP_DIALOG_CAP_SUPPORTED; + if (rem_can_update) { + status = pjsip_inv_update(call->inv, NULL, new_sdp, &tdata); + } else { + status = pjsip_inv_reinvite(call->inv, NULL, new_sdp, &tdata); + } + + if (status==PJ_EINVALIDOP && + ++call->lock_codec.retry_cnt <= LOCK_CODEC_MAX_RETRY) + { + /* Ups, let's reschedule again */ + pj_time_val delay = {0, LOCK_CODEC_RETRY_INTERVAL}; + pj_time_val_normalize(&delay); + call->lock_codec.reinv_timer.id = PJ_TRUE; + pjsip_endpt_schedule_timer(pjsua_var.endpt, + &call->lock_codec.reinv_timer, &delay); + return status; + } else if (status != PJ_SUCCESS) { + pjsua_perror(THIS_FILE, "Error creating UPDATE/re-INVITE to lock codec", + status); + return status; + } + + /* Send the UPDATE/re-INVITE request */ + status = pjsip_inv_send_msg(call->inv, tdata); + if (status != PJ_SUCCESS) { + pjsua_perror(THIS_FILE, "Error sending UPDATE/re-INVITE in lock codec", + status); + return status; + } + + return status; +} + +/* Check if remote answerer has given us more than one codecs. If so, + * create another offer with one codec only to lock down the codec. + */ +static pj_status_t lock_codec(pjsua_call *call) +{ + pjsip_inv_session *inv = call->inv; + const pjmedia_sdp_session *local_sdp, *remote_sdp; + pj_time_val delay = {0, 0}; + const pj_str_t st_update = {"UPDATE", 6}; + unsigned i; + pj_bool_t has_mult_fmt = PJ_FALSE; + pj_status_t status; + + /* Stop lock codec timer, if it is active */ + if (call->lock_codec.reinv_timer.id) { + pjsip_endpt_cancel_timer(pjsua_var.endpt, + &call->lock_codec.reinv_timer); + call->lock_codec.reinv_timer.id = PJ_FALSE; + } + + /* Skip this if we are the answerer */ + if (!inv->neg || !pjmedia_sdp_neg_was_answer_remote(inv->neg)) { + return PJ_SUCCESS; + } + + /* 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->lock_codec.pending = PJ_TRUE; + return PJ_SUCCESS; + } + + status = pjmedia_sdp_neg_get_active_local(inv->neg, &local_sdp); + if (status != PJ_SUCCESS) + return status; + status = pjmedia_sdp_neg_get_active_remote(inv->neg, &remote_sdp); + if (status != PJ_SUCCESS) + return status; + + /* Find multiple codecs answer in all media */ + for (i = 0; i < call->med_cnt; ++i) { + pjsua_call_media *call_med = &call->media[i]; + const pjmedia_sdp_media *rem_m, *loc_m; + unsigned codec_cnt = 0; + + /* 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); + + /* Count the formats in the answer. */ + if (rem_m->desc.fmt_count==1) { + codec_cnt = 1; + } else { + unsigned j; + for (j=0; j<rem_m->desc.fmt_count && codec_cnt <= 1; ++j) { + if (!is_non_av_fmt(rem_m, &rem_m->desc.fmt[j])) + ++codec_cnt; + } + } + + if (codec_cnt > 1) { + has_mult_fmt = PJ_TRUE; + break; + } + } + + /* Each media in the answer already contains single codec. */ + if (!has_mult_fmt) { + call->lock_codec.retry_cnt = 0; + return PJ_SUCCESS; + } + + /* Remote keeps answering with multiple codecs, let's just give up + * locking codec to avoid infinite retry loop. + */ + if (++call->lock_codec.retry_cnt > LOCK_CODEC_MAX_RETRY) + return PJ_SUCCESS; + + PJ_LOG(4, (THIS_FILE, "Got answer with multiple codecs, scheduling " + "updating media session to use only one codec..")); + + call->lock_codec.sdp_ver = local_sdp->origin.version; + + /* Can't send UPDATE or re-INVITE now, so just schedule it immediately. + * See: https://trac.pjsip.org/repos/ticket/1149 + */ + pj_timer_entry_init(&call->lock_codec.reinv_timer, PJ_TRUE, + (void*)(pj_size_t)call->index, + &reinv_timer_cb); + pjsip_endpt_schedule_timer(pjsua_var.endpt, + &call->lock_codec.reinv_timer, &delay); + + 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; + + 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 lock codec was pended as media update was done in the + * EARLY state and remote does not support UPDATE. + */ + if (call->lock_codec.pending) { + pj_status_t status; + status = lock_codec(call); + if (status != PJ_SUCCESS) { + pjsua_perror(THIS_FILE, "Unable to lock codec", status); + } + call->lock_codec.pending = PJ_FALSE; + } + 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 lock codec timer, if it is active */ + if (call->lock_codec.reinv_timer.id) { + pjsip_endpt_cancel_timer(pjsua_var.endpt, + &call->lock_codec.reinv_timer); + call->lock_codec.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); + } + } + } + } + + + if (pjsua_var.ua_cfg.cb.on_call_state) + (*pjsua_var.ua_cfg.cb.on_call_state)(call->index, e); + + /* 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_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 ) +{ + pjsua_call *call; + pjsip_tx_data *tdata; + pj_status_t status; + + call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id]; + + status = pjsip_inv_end_session(inv, code, NULL, &tdata); + if (status != PJ_SUCCESS) + return; + + /* Add SDP in 488 status */ +#if DISABLED_FOR_TICKET_1185 + 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 (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; + } + + /* 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 (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. */ + status = lock_codec(call); + if (status != PJ_SUCCESS) { + pjsua_perror(THIS_FILE, "Unable to lock codec", status); + } + + /* 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) +{ + 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; mi<sdp->media_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"); + + if (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); + 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, + const pjmedia_sdp_session *offer) +{ + pjsua_call *call; + pjmedia_sdp_session *answer; + unsigned i; + pj_status_t status; + + 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_var.ua_cfg.cb.on_call_rx_offer) { + pjsip_status_code code = PJSIP_SC_OK; + pjsua_call_setting opt = call->opt; + + (*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; + } + + /* 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); + } + + 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 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; + + pj_log_push_indent(); + + call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id]; + + /* 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 + * transfered 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, 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; + } + + /* This better be a NOTIFY request */ + if (event->type == PJSIP_EVENT_TSX_STATE && + event->body.tsx_state.type == PJSIP_EVENT_RX_MSG) + { + pjsip_rx_data *rdata; + + rdata = event->body.tsx_state.src.rdata; + + /* Check if there's body */ + msg = rdata->msg_info.msg; + body = msg->body; + if (!body) { + 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 = 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_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_transfered( 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); + } + + 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 transfered 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 *tdata; + const pj_str_t str_false = { "false", 5}; + pjsip_hdr *hdr; + + status = pjsip_dlg_create_response(inv->dlg, rdata, code, NULL, + &tdata); + 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(tdata->pool, &str_refer_sub, + &str_false); + pjsip_msg_add_hdr(tdata->msg, hdr); + + + /* Send answer */ + status = pjsip_dlg_send_response(inv->dlg, pjsip_rdata_get_tsx(rdata), + tdata); + 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) { + /* Shouldn't happen. It happens only when we don't terminate the + * server subscription caused by REFER after the call has been + * transfered (and this call has been disconnected), and we + * receive another REFER for this call. + */ + 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_transfered(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_msg *msg; + pjsip_accept_hdr *accept_hdr; + pj_status_t status; + + rdata = e->body.tsx_state.src.rdata; + msg = rdata->msg_info.msg; + + /* 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. + */ + status = 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 && + tsx->last_tx == (pjsip_tx_data*)call->hold_msg && + tsx->state >= PJSIP_TSX_STATE_COMPLETED) + { + /* Monitor the status of call hold request */ + call->hold_msg = NULL; + if (tsx->status_code/100 != 2) { + /* 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 (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 }; + 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); + } + } + } + +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; +} + |