* #27232: jni: added pjproject checkout as regular git content
We will remove it once the next release of pjsip (with Android support)
comes out and is merged into SFLphone.
diff --git a/jni/pjproject-android/.svn/pristine/73/7305e7d1ae7a1a8f5ef251004243552fa18abd8a.svn-base b/jni/pjproject-android/.svn/pristine/73/7305e7d1ae7a1a8f5ef251004243552fa18abd8a.svn-base
new file mode 100644
index 0000000..09c7645
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/73/7305e7d1ae7a1a8f5ef251004243552fa18abd8a.svn-base
@@ -0,0 +1 @@
+#include "../../portaudio/src/os/win/pa_win_hostapis.c"
diff --git a/jni/pjproject-android/.svn/pristine/73/732fada170dae9e07488070226e3e68f1e49a05c.svn-base b/jni/pjproject-android/.svn/pristine/73/732fada170dae9e07488070226e3e68f1e49a05c.svn-base
new file mode 100644
index 0000000..7d19f6f
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/73/732fada170dae9e07488070226e3e68f1e49a05c.svn-base
@@ -0,0 +1,11 @@
+# $Id$
+#
+from inc_cfg import *
+
+# simple test
+test_param = TestParam(
+ "Resample (large filter) 8 KHZ to 11 KHZ",
+ [
+ InstanceParam("endpt", "--null-audio --quality 10 --clock-rate 11025 --play-file wavs/input.8.wav --rec-file wavs/tmp.11.wav")
+ ]
+ )
diff --git a/jni/pjproject-android/.svn/pristine/73/736e296e1d08ba948a69db732914d772501b566a.svn-base b/jni/pjproject-android/.svn/pristine/73/736e296e1d08ba948a69db732914d772501b566a.svn-base
new file mode 100644
index 0000000..35571e2
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/73/736e296e1d08ba948a69db732914d772501b566a.svn-base
Binary files differ
diff --git a/jni/pjproject-android/.svn/pristine/73/7375d283e50015a03a261a257197b26a1f764717.svn-base b/jni/pjproject-android/.svn/pristine/73/7375d283e50015a03a261a257197b26a1f764717.svn-base
new file mode 100644
index 0000000..c591823
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/73/7375d283e50015a03a261a257197b26a1f764717.svn-base
Binary files differ
diff --git a/jni/pjproject-android/.svn/pristine/73/73a21d3d0947020f3ce205deda26ea697544e2f1.svn-base b/jni/pjproject-android/.svn/pristine/73/73a21d3d0947020f3ce205deda26ea697544e2f1.svn-base
new file mode 100644
index 0000000..4f36dd9
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/73/73a21d3d0947020f3ce205deda26ea697544e2f1.svn-base
@@ -0,0 +1,20 @@
+<?xml version="1.0" ?>
+<Scenario site="$(HOSTNAME)" url="http://my.cdash.org/submit.php?project=PJSIP" wdir="$(PJDIR)">
+ <!-- Symbian S60 default -->
+ <Submit group="Experimental" build="$(OS)-$(S60TARGETNAME)-default" >
+ <Update disabled="1" />
+ <FileWrite file="pjlib/include/pj/config_site.h">
+ <![CDATA[
+/* Written by ccdash */
+#include <pj/config_site_sample.h>
+#undef PJ_HAS_IPV6
+#define PJ_HAS_IPV6 1
+#undef PJMEDIA_HAS_G7221_CODEC
+#define PJMEDIA_HAS_G7221_CODEC 1
+]]>
+ </FileWrite>
+ <Configure cmd="cmd /c devices -setdefault $(S60DEVICE)" />
+ <Build wdir="build.symbian" cmd='cmd /C "SET EPOCROOT=$(EPOCROOT)&& bldmake bldfiles && abld reallyclean $(S60TARGET) && abld build $(S60TARGET)"' />
+ </Submit>
+</Scenario>
+
diff --git a/jni/pjproject-android/.svn/pristine/73/73ac2e77e9bae5df0a0c457b65f491df7efa5fde.svn-base b/jni/pjproject-android/.svn/pristine/73/73ac2e77e9bae5df0a0c457b65f491df7efa5fde.svn-base
new file mode 100644
index 0000000..1cd91fc
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/73/73ac2e77e9bae5df0a0c457b65f491df7efa5fde.svn-base
@@ -0,0 +1,4645 @@
+/* $Id$ */
+/*
+ * 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);
+
+/* Timer callback to send re-INVITE/UPDATE to lock codec or ICE update */
+static void reinv_timer_cb(pj_timer_heap_t *th, pj_timer_entry *entry);
+
+/* Check and send reinvite for lock codec and ICE update */
+static pj_status_t process_pending_reinvite(pjsua_call *call);
+
+/*
+ * Reset call descriptor.
+ */
+static void reset_call(pjsua_call_id id)
+{
+ pjsua_call *call = &pjsua_var.calls[id];
+ unsigned i;
+
+ 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);
+ pj_timer_entry_init(&call->reinv_timer, PJ_FALSE,
+ (void*)(pj_size_t)id, &reinv_timer_cb);
+}
+
+
+/*
+ * 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);
+
+ /* Add "INFO" in Allow header, for DTMF and video key frame request. */
+ pjsip_endpt_add_capability(pjsua_var.endpt, NULL, PJSIP_H_ALLOW,
+ NULL, 1, &pjsip_info_method.name);
+
+ 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_bool_t cb_called = PJ_FALSE;
+ pj_status_t status = (info? info->status: PJ_SUCCESS);
+
+ PJSUA_LOCK();
+
+ /* Increment the dialog's lock otherwise when invite session creation
+ * fails the dialog will be destroyed prematurely.
+ */
+ pjsip_dlg_inc_lock(dlg);
+
+ /* Decrement dialog session. */
+ pjsip_dlg_dec_session(dlg, &pjsua_var.mod);
+
+ if (status != PJ_SUCCESS) {
+ pj_str_t err_str;
+ pj_ssize_t title_len;
+
+ call->last_code = PJSIP_SC_TEMPORARILY_UNAVAILABLE;
+ pj_strcpy2(&call->last_text, "Media init error: ");
+
+ title_len = call->last_text.slen;
+ err_str = pj_strerror(status, call->last_text_buf_ + title_len,
+ sizeof(call->last_text_buf_) - title_len);
+ call->last_text.slen += err_str.slen;
+
+ pjsua_perror(THIS_FILE, "Error initializing media channel", status);
+ goto on_error;
+ }
+
+ /* pjsua_media_channel_deinit() has been called or
+ * call has been hung up.
+ */
+ if (call->async_call.med_ch_deinit ||
+ call->async_call.call_var.out_call.hangup)
+ {
+ PJ_LOG(4,(THIS_FILE, "Call has been hung up or media channel has "
+ "been deinitialized"));
+ goto on_error;
+ }
+
+ /* Create offer */
+ 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) {
+ cb_called = PJ_TRUE;
+
+ /* Upon failure to send first request, the invite
+ * session would have been cleared.
+ */
+ inv = NULL;
+ goto on_error;
+ }
+
+ /* Done. */
+ call->med_ch_cb = NULL;
+
+ pjsip_dlg_dec_lock(dlg);
+ PJSUA_UNLOCK();
+
+ return PJ_SUCCESS;
+
+on_error:
+ if (inv == NULL && call_id != -1 && !cb_called &&
+ pjsua_var.ua_cfg.cb.on_call_state)
+ {
+ (*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) {
+ pjsua_media_channel_deinit(call_id);
+ reset_call(call_id);
+ }
+
+ call->med_ch_cb = NULL;
+
+ pjsua_check_snd_dev_idle();
+
+ 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
+
+ call->opt = *opt;
+
+ /* If call is established, reinit media channel */
+ if (call->inv && call->inv->state == PJSIP_INV_STATE_CONFIRMED) {
+ pjsip_role_e role = rem_sdp? PJSIP_ROLE_UAS : PJSIP_ROLE_UAC;
+ pj_status_t status;
+
+ status = pjsua_media_channel_init(call->index, role,
+ call->secure_level,
+ call->inv->pool_prov,
+ rem_sdp, NULL,
+ PJ_FALSE, NULL);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Error re-initializing media channel",
+ status);
+ return status;
+ }
+ }
+
+ return PJ_SUCCESS;
+}
+
+/*
+ * 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;
+ }
+
+ /* Clear call descriptor */
+ reset_call(call_id);
+
+ call = &pjsua_var.calls[call_id];
+
+ /* Associate session with account */
+ call->acc_id = acc_id;
+ call->call_hold_type = acc->cfg.call_hold_type;
+
+ /* 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,
+ (msg_data && msg_data->target_uri.slen?
+ &msg_data->target_uri: dest_uri),
+ &dlg);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Dialog creation failed", status);
+ goto on_error;
+ }
+
+ /* Increment the dialog's lock otherwise when invite session creation
+ * fails the dialog will be destroyed prematurely.
+ */
+ pjsip_dlg_inc_lock(dlg);
+
+ 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) {
+ pjsua_media_channel_deinit(call_id);
+ reset_call(call_id);
+ }
+
+ pjsua_check_snd_dev_idle();
+
+ if (tmp_pool)
+ pj_pool_release(tmp_pool);
+ PJSUA_UNLOCK();
+
+ pj_log_pop_indent();
+ return status;
+}
+
+
+/* Get the NAT type information in remote's SDP */
+static void update_remote_nat_type(pjsua_call *call,
+ const pjmedia_sdp_session *sdp)
+{
+ const pjmedia_sdp_attr *xnat;
+
+ xnat = pjmedia_sdp_attr_find2(sdp->attr_count, sdp->attr, "X-nat", NULL);
+ if (xnat) {
+ call->rem_nat_type = (pj_stun_nat_type) (xnat->value.ptr[0] - '0');
+ } else {
+ call->rem_nat_type = PJ_STUN_NAT_TYPE_UNKNOWN;
+ }
+
+ PJ_LOG(5,(THIS_FILE, "Call %d: remote NAT type is %d (%s)", call->index,
+ call->rem_nat_type, pj_stun_get_nat_name(call->rem_nat_type)));
+}
+
+
+static pj_status_t process_incoming_call_replace(pjsua_call *call,
+ pjsip_dialog *replaced_dlg)
+{
+ pjsip_inv_session *replaced_inv;
+ struct pjsua_call *replaced_call;
+ pjsip_tx_data *tdata = NULL;
+ pj_status_t status = PJ_SUCCESS;
+
+ /* Get the invite session in the dialog */
+ replaced_inv = pjsip_dlg_get_inv_session(replaced_dlg);
+
+ /* Get the replaced call instance */
+ replaced_call = (pjsua_call*) replaced_dlg->mod_data[pjsua_var.mod.id];
+
+ /* Notify application */
+ if (pjsua_var.ua_cfg.cb.on_call_replaced)
+ pjsua_var.ua_cfg.cb.on_call_replaced(replaced_call->index,
+ call->index);
+
+ if (replaced_call->inv->state <= PJSIP_INV_STATE_EARLY &&
+ replaced_call->inv->role != PJSIP_ROLE_UAC)
+ {
+ if (replaced_call->last_code > 100 && replaced_call->last_code < 200)
+ {
+ pjsip_status_code code = replaced_call->last_code;
+ pj_str_t *text = &replaced_call->last_text;
+
+ PJ_LOG(4,(THIS_FILE, "Answering replacement call %d with %d/%.*s",
+ call->index, code, text->slen, text->ptr));
+
+ /* Answer the new call with last response in the replaced call */
+ status = pjsip_inv_answer(call->inv, code, text, NULL, &tdata);
+ }
+ } else {
+ PJ_LOG(4,(THIS_FILE, "Answering replacement call %d with 200/OK",
+ call->index));
+
+ /* Answer the new call with 200 response */
+ status = pjsip_inv_answer(call->inv, 200, NULL, NULL, &tdata);
+ }
+
+ if (status == PJ_SUCCESS && tdata)
+ status = pjsip_inv_send_msg(call->inv, tdata);
+
+ if (status != PJ_SUCCESS)
+ pjsua_perror(THIS_FILE, "Error answering session", status);
+
+ /* Note that inv may be invalid if 200/OK has caused error in
+ * starting the media.
+ */
+
+ PJ_LOG(4,(THIS_FILE, "Disconnecting replaced call %d",
+ replaced_call->index));
+
+ /* Disconnect replaced invite session */
+ status = pjsip_inv_end_session(replaced_inv, PJSIP_SC_GONE, NULL,
+ &tdata);
+ if (status == PJ_SUCCESS && tdata)
+ status = pjsip_inv_send_msg(replaced_inv, tdata);
+
+ if (status != PJ_SUCCESS)
+ pjsua_perror(THIS_FILE, "Error terminating session", status);
+
+ return status;
+}
+
+
+static void process_pending_call_answer(pjsua_call *call)
+{
+ struct call_answer *answer, *next;
+
+ 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 = PJSIP_SC_INTERNAL_SERVER_ERROR;
+ 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.acc[acc_id].cfg.ice_cfg.enable_ice)
+ options |= PJSIP_INV_SUPPORT_ICE;
+ if (pjsua_var.acc[acc_id].cfg.use_timer == PJSUA_SIP_TIMER_REQUIRED)
+ options |= PJSIP_INV_REQUIRE_TIMER;
+ else if (pjsua_var.acc[acc_id].cfg.use_timer == PJSUA_SIP_TIMER_ALWAYS)
+ options |= PJSIP_INV_ALWAYS_USE_TIMER;
+
+ status = pjsip_inv_verify_request2(rdata, &options, offer, NULL, NULL,
+ pjsua_var.endpt, &response);
+ if (status != PJ_SUCCESS) {
+
+ /*
+ * No we can't handle the incoming INVITE request.
+ */
+ if (response) {
+ pjsip_response_addr res_addr;
+
+ pjsip_get_response_addr(response->pool, rdata, &res_addr);
+ pjsip_endpt_send_response(pjsua_var.endpt, &res_addr, response,
+ NULL, NULL);
+
+ } else {
+ /* Respond with 500 (Internal Server Error) */
+ pjsip_endpt_respond(pjsua_var.endpt, NULL, rdata, 500, NULL,
+ NULL, NULL, NULL);
+ }
+
+ goto on_return;
+ }
+
+ /* Get suitable Contact header */
+ if (pjsua_var.acc[acc_id].contact.slen) {
+ contact = pjsua_var.acc[acc_id].contact;
+ } else {
+ status = pjsua_acc_create_uas_contact(rdata->tp_info.pool, &contact,
+ acc_id, rdata);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Unable to generate Contact header",
+ status);
+ pjsip_endpt_respond_stateless(pjsua_var.endpt, rdata, 500, NULL,
+ NULL, NULL);
+ goto on_return;
+ }
+ }
+
+ /* Create dialog: */
+ status = pjsip_dlg_create_uas( 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;
+ call->async_call.dlg = 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;
+ call->async_call.dlg = NULL;
+ goto on_return;
+ }
+ }
+
+ /* Create answer */
+/*
+ status = pjsua_media_channel_create_sdp(call->index, rdata->tp_info.pool,
+ offer, &answer, &sip_err_code);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Error creating SDP answer", status);
+ pjsip_endpt_respond(pjsua_var.endpt, NULL, rdata,
+ sip_err_code, NULL, NULL, NULL, NULL);
+ goto on_return;
+ }
+*/
+
+ /* Init Session Timers */
+ status = pjsip_timer_init_session(inv,
+ &pjsua_var.acc[acc_id].cfg.timer_setting);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Session Timer init failed", status);
+ pjsip_dlg_respond(dlg, rdata, PJSIP_SC_INTERNAL_SERVER_ERROR, NULL, NULL, NULL);
+ pjsip_inv_terminate(inv, PJSIP_SC_INTERNAL_SERVER_ERROR, PJ_FALSE);
+
+ pjsua_media_channel_deinit(call->index);
+ call->inv = NULL;
+ call->async_call.dlg = NULL;
+
+ goto on_return;
+ }
+
+ /* Update NAT type of remote endpoint, only when there is SDP in
+ * incoming INVITE!
+ */
+ if (pjsua_var.ua_cfg.nat_type_in_sdp && inv->neg &&
+ pjmedia_sdp_neg_get_state(inv->neg) > PJMEDIA_SDP_NEG_STATE_LOCAL_OFFER)
+ {
+ const pjmedia_sdp_session *remote_sdp;
+
+ if (pjmedia_sdp_neg_get_neg_remote(inv->neg, &remote_sdp)==PJ_SUCCESS)
+ update_remote_nat_type(call, remote_sdp);
+ }
+
+ /* Must answer with some response to initial INVITE. We'll do this before
+ * attaching the call to the invite session/dialog, so that the application
+ * will not get notification about this event (on another scenario, it is
+ * also possible that inv_send_msg() fails and causes the invite session to
+ * be disconnected. If we have the call attached at this time, this will
+ * cause the disconnection callback to be called before on_incoming_call()
+ * callback is called, which is not right).
+ */
+ status = pjsip_inv_initial_answer(inv, rdata,
+ 100, NULL, NULL, &response);
+ if (status != PJ_SUCCESS) {
+ if (response == NULL) {
+ pjsua_perror(THIS_FILE, "Unable to send answer to incoming INVITE",
+ status);
+ pjsip_dlg_respond(dlg, rdata, 500, NULL, NULL, NULL);
+ pjsip_inv_terminate(inv, 500, PJ_FALSE);
+ } else {
+ pjsip_inv_send_msg(inv, response);
+ pjsip_inv_terminate(inv, response->msg->line.status.code,
+ PJ_FALSE);
+ }
+ pjsua_media_channel_deinit(call->index);
+ call->inv = NULL;
+ call->async_call.dlg = NULL;
+ goto on_return;
+
+ } else {
+ status = pjsip_inv_send_msg(inv, response);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Unable to send 100 response", status);
+ pjsua_media_channel_deinit(call->index);
+ call->inv = NULL;
+ call->async_call.dlg = NULL;
+ goto on_return;
+ }
+ }
+
+ /* Only do this after sending 100/Trying (really! see the long comment
+ * above)
+ */
+ 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;
+ pjsip_dialog *dlg = NULL;
+
+ pj_gettimeofday(&time_start);
+ timeout.sec = 0;
+ timeout.msec = PJSUA_ACQUIRE_CALL_TIMEOUT;
+ pj_time_val_normalize(&timeout);
+
+ for (retry=0; ; ++retry) {
+
+ if (retry % 10 == 9) {
+ pj_time_val dtime;
+
+ pj_gettimeofday(&dtime);
+ PJ_TIME_VAL_SUB(dtime, time_start);
+ if (!PJ_TIME_VAL_LT(dtime, timeout))
+ break;
+ }
+
+ has_pjsua_lock = PJ_FALSE;
+
+ status = PJSUA_TRY_LOCK();
+ if (status != PJ_SUCCESS) {
+ pj_thread_sleep(retry/10);
+ continue;
+ }
+
+ has_pjsua_lock = PJ_TRUE;
+ call = &pjsua_var.calls[call_id];
+ if (call->inv)
+ dlg = call->inv->dlg;
+ else
+ dlg = call->async_call.dlg;
+
+ if (dlg == NULL) {
+ PJSUA_UNLOCK();
+ PJ_LOG(3,(THIS_FILE, "Invalid call_id %d in %s", call_id, title));
+ return PJSIP_ESESSIONTERMINATED;
+ }
+
+ status = pjsip_dlg_try_inc_lock(dlg);
+ if (status != PJ_SUCCESS) {
+ PJSUA_UNLOCK();
+ pj_thread_sleep(retry/10);
+ continue;
+ }
+
+ PJSUA_UNLOCK();
+
+ break;
+ }
+
+ if (status != PJ_SUCCESS) {
+ if (has_pjsua_lock == PJ_FALSE)
+ PJ_LOG(1,(THIS_FILE, "Timed-out trying to acquire PJSUA mutex "
+ "(possibly system has deadlocked) in %s",
+ title));
+ else
+ PJ_LOG(1,(THIS_FILE, "Timed-out trying to acquire dialog mutex "
+ "(possibly system has deadlocked) in %s",
+ title));
+ return PJ_ETIMEDOUT;
+ }
+
+ *p_call = call;
+ *p_dlg = dlg;
+
+ return PJ_SUCCESS;
+}
+
+
+/*
+ * Obtain detail information about the specified call.
+ */
+PJ_DEF(pj_status_t) pjsua_call_get_info( pjsua_call_id call_id,
+ pjsua_call_info *info)
+{
+ pjsua_call *call;
+ pjsip_dialog *dlg;
+ unsigned mi;
+
+ PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
+ PJ_EINVAL);
+
+ pj_bzero(info, sizeof(*info));
+
+ /* Use PJSUA_LOCK() instead of acquire_call():
+ * https://trac.pjsip.org/repos/ticket/1371
+ */
+ PJSUA_LOCK();
+
+ call = &pjsua_var.calls[call_id];
+ dlg = (call->inv ? call->inv->dlg : call->async_call.dlg);
+ if (!dlg) {
+ PJSUA_UNLOCK();
+ return PJSIP_ESESSIONTERMINATED;
+ }
+
+ /* id and role */
+ info->id = call_id;
+ info->role = dlg->role;
+ info->acc_id = call->acc_id;
+
+ /* local info */
+ info->local_info.ptr = info->buf_.local_info;
+ pj_strncpy(&info->local_info, &dlg->local.info_str,
+ sizeof(info->buf_.local_info));
+
+ /* local contact */
+ info->local_contact.ptr = info->buf_.local_contact;
+ info->local_contact.slen = pjsip_uri_print(PJSIP_URI_IN_CONTACT_HDR,
+ dlg->local.contact->uri,
+ info->local_contact.ptr,
+ sizeof(info->buf_.local_contact));
+
+ /* 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 media transport creation is not yet completed, we will hangup
+ * the call in the media transport creation callback instead.
+ */
+ if (call->med_ch_cb && !call->inv) {
+ PJ_LOG(4,(THIS_FILE, "Pending call %d hangup upon completion "
+ "of media transport", call_id));
+ call->async_call.call_var.out_call.hangup = PJ_TRUE;
+ if (code == 0)
+ call->last_code = PJSIP_SC_REQUEST_TERMINATED;
+ else
+ call->last_code = (pjsip_status_code)code;
+ if (reason) {
+ pj_strncpy(&call->last_text, reason,
+ sizeof(call->last_text_buf_));
+ }
+
+ goto on_return;
+ }
+
+ if (code==0) {
+ if (call->inv->state == PJSIP_INV_STATE_CONFIRMED)
+ code = PJSIP_SC_OK;
+ else if (call->inv->role == PJSIP_ROLE_UAS)
+ code = PJSIP_SC_DECLINE;
+ else
+ code = PJSIP_SC_REQUEST_TERMINATED;
+ }
+
+ status = pjsip_inv_end_session(call->inv, code, reason, &tdata);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE,
+ "Failed to create end session message",
+ status);
+ 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 reinvite timer, if it is active */
+ if (call->reinv_timer.id) {
+ pjsua_cancel_timer(&call->reinv_timer);
+ call->reinv_timer.id = PJ_FALSE;
+ }
+
+on_return:
+ if (dlg) pjsip_dlg_dec_lock(dlg);
+ pj_log_pop_indent();
+ return status;
+}
+
+
+/*
+ * Accept or reject redirection.
+ */
+PJ_DEF(pj_status_t) pjsua_call_process_redirect( pjsua_call_id call_id,
+ pjsip_redirect_op cmd)
+{
+ pjsua_call *call;
+ pjsip_dialog *dlg;
+ pj_status_t status;
+
+ PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
+ PJ_EINVAL);
+
+ status = acquire_call("pjsua_call_process_redirect()", call_id,
+ &call, &dlg);
+ if (status != PJ_SUCCESS)
+ return status;
+
+ status = pjsip_inv_process_redirect(call->inv, cmd, NULL);
+
+ pjsip_dlg_dec_lock(dlg);
+
+ return status;
+}
+
+
+/*
+ * Put the specified call on hold.
+ */
+PJ_DEF(pj_status_t) pjsua_call_set_hold(pjsua_call_id call_id,
+ const pjsua_msg_data *msg_data)
+{
+ return pjsua_call_set_hold2(call_id, 0, msg_data);
+}
+
+PJ_DEF(pj_status_t) pjsua_call_set_hold2(pjsua_call_id call_id,
+ unsigned options,
+ const pjsua_msg_data *msg_data)
+{
+ pjmedia_sdp_session *sdp;
+ pjsua_call *call;
+ pjsip_dialog *dlg = NULL;
+ pjsip_tx_data *tdata;
+ pj_str_t *new_contact = NULL;
+ pj_status_t status;
+
+ PJ_ASSERT_RETURN(call_id>=0 && call_id<(int)pjsua_var.ua_cfg.max_calls,
+ PJ_EINVAL);
+
+ PJ_LOG(4,(THIS_FILE, "Putting call %d on hold", call_id));
+ pj_log_push_indent();
+
+ status = acquire_call("pjsua_call_set_hold()", call_id, &call, &dlg);
+ if (status != PJ_SUCCESS)
+ goto on_return;
+
+ if (call->inv->state != PJSIP_INV_STATE_CONFIRMED) {
+ PJ_LOG(3,(THIS_FILE, "Can not hold call that is not confirmed"));
+ status = PJSIP_ESESSIONSTATE;
+ goto on_return;
+ }
+
+ status = create_sdp_of_call_hold(call, &sdp);
+ if (status != PJ_SUCCESS)
+ goto on_return;
+
+ if ((options & PJSUA_CALL_UPDATE_CONTACT) &&
+ pjsua_acc_is_valid(call->acc_id))
+ {
+ 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);
+
+ /* 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 */
+ 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 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;
+ }
+
+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();
+}
+
+
+/* Timer callback to send re-INVITE/UPDATE to lock codec or ICE update */
+static void reinv_timer_cb(pj_timer_heap_t *th, pj_timer_entry *entry)
+{
+ 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].reinv_timer.id = PJ_FALSE;
+
+ pj_log_push_indent();
+
+ status = acquire_call("reinv_timer_cb()", call_id, &call, &dlg);
+ if (status != PJ_SUCCESS) {
+ pj_log_pop_indent();
+ return;
+ }
+
+ process_pending_reinvite(call);
+
+ pjsip_dlg_dec_lock(dlg);
+
+ pj_log_pop_indent();
+}
+
+
+/* Check if the specified format can be skipped in counting codecs */
+static pj_bool_t is_non_av_fmt(const pjmedia_sdp_media *m,
+ const pj_str_t *fmt)
+{
+ const pj_str_t STR_TEL = {"telephone-event", 15};
+ unsigned pt;
+
+ pt = pj_strtoul(fmt);
+
+ /* Check for comfort noise */
+ if (pt == PJMEDIA_RTP_PT_CN)
+ return PJ_TRUE;
+
+ /* Dynamic PT, check the format name */
+ if (pt >= 96) {
+ pjmedia_sdp_attr *a;
+ pjmedia_sdp_rtpmap rtpmap;
+
+ /* Get the format name */
+ a = pjmedia_sdp_attr_find2(m->attr_count, m->attr, "rtpmap", fmt);
+ if (a && pjmedia_sdp_attr_get_rtpmap(a, &rtpmap)==PJ_SUCCESS) {
+ /* Check for telephone-event */
+ if (pj_stricmp(&rtpmap.enc_name, &STR_TEL)==0)
+ return PJ_TRUE;
+ } else {
+ /* Invalid SDP, should not reach here */
+ pj_assert(!"SDP should have been validated!");
+ return PJ_TRUE;
+ }
+ }
+
+ return PJ_FALSE;
+}
+
+
+/* Schedule check for the need of re-INVITE/UPDATE after media update, cases:
+ * - lock codec if remote answerer has given us more than one codecs
+ * - update ICE default transport address if it has changed after ICE
+ * connectivity check.
+ */
+void pjsua_call_schedule_reinvite_check(pjsua_call *call, unsigned delay_ms)
+{
+ pj_time_val delay;
+
+ /* Stop reinvite timer, if it is active */
+ if (call->reinv_timer.id)
+ pjsua_cancel_timer(&call->reinv_timer);
+
+ delay.sec = 0;
+ delay.msec = delay_ms;
+ pj_time_val_normalize(&delay);
+ call->reinv_timer.id = PJ_TRUE;
+ pjsua_schedule_timer(&call->reinv_timer, &delay);
+}
+
+
+/* Check if lock codec is needed */
+static pj_bool_t check_lock_codec(pjsua_call *call)
+{
+ const pjmedia_sdp_session *local_sdp, *remote_sdp;
+ pj_bool_t has_mult_fmt = PJ_FALSE;
+ unsigned i;
+ pj_status_t status;
+
+ /* Check if lock codec is disabled */
+ if (!pjsua_var.acc[call->acc_id].cfg.lock_codec)
+ return PJ_FALSE;
+
+ /* Check lock codec retry count */
+ if (call->lock_codec.retry_cnt >= LOCK_CODEC_MAX_RETRY)
+ return PJ_FALSE;
+
+ /* Check if we are the answerer, we shouldn't need to lock codec */
+ if (!call->inv->neg || !pjmedia_sdp_neg_was_answer_remote(call->inv->neg))
+ return PJ_FALSE;
+
+ /* Check if remote answerer has given us more than one codecs. */
+ status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &local_sdp);
+ if (status != PJ_SUCCESS)
+ return PJ_FALSE;
+ status = pjmedia_sdp_neg_get_active_remote(call->inv->neg, &remote_sdp);
+ if (status != PJ_SUCCESS)
+ return PJ_FALSE;
+
+ for (i = 0; i < call->med_cnt && !has_mult_fmt; ++i) {
+ pjsua_call_media *call_med = &call->media[i];
+ const pjmedia_sdp_media *rem_m, *loc_m;
+ unsigned codec_cnt = 0;
+ unsigned j;
+
+ /* Skip this if the media is inactive or error */
+ if (call_med->state == PJSUA_CALL_MEDIA_NONE ||
+ call_med->state == PJSUA_CALL_MEDIA_ERROR ||
+ call_med->dir == PJMEDIA_DIR_NONE)
+ {
+ continue;
+ }
+
+ /* Remote may answer with less media lines. */
+ if (i >= remote_sdp->media_count)
+ continue;
+
+ rem_m = remote_sdp->media[i];
+ loc_m = local_sdp->media[i];
+
+ /* Verify that media must be active. */
+ pj_assert(loc_m->desc.port && rem_m->desc.port);
+
+ /* Count the formats in the answer. */
+ 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 > 1)
+ has_mult_fmt = PJ_TRUE;
+ }
+ }
+
+ /* Reset retry count when remote answer has one codec */
+ if (!has_mult_fmt)
+ call->lock_codec.retry_cnt = 0;
+
+ return has_mult_fmt;
+}
+
+/* Check if ICE setup is complete and if it needs to send reinvite */
+static pj_bool_t check_ice_complete(pjsua_call *call, pj_bool_t *need_reinv)
+{
+ pj_bool_t ice_need_reinv = PJ_FALSE;
+ pj_bool_t ice_complete = PJ_TRUE;
+ unsigned i;
+
+ /* Check if ICE setup is complete and if it needs reinvite */
+ for (i = 0; i < call->med_cnt; ++i) {
+ pjsua_call_media *call_med = &call->media[i];
+ pjmedia_transport_info tpinfo;
+ pjmedia_ice_transport_info *ice_info;
+
+ if (call_med->tp_st == PJSUA_MED_TP_NULL ||
+ call_med->tp_st == PJSUA_MED_TP_DISABLED ||
+ call_med->state == PJSUA_CALL_MEDIA_ERROR)
+ {
+ continue;
+ }
+
+ pjmedia_transport_info_init(&tpinfo);
+ pjmedia_transport_get_info(call_med->tp, &tpinfo);
+ ice_info = (pjmedia_ice_transport_info*)
+ pjmedia_transport_info_get_spc_info(
+ &tpinfo, PJMEDIA_TRANSPORT_TYPE_ICE);
+
+ /* Check if ICE is active */
+ if (!ice_info || !ice_info->active)
+ continue;
+
+ /* Check if ICE setup not completed yet */
+ if (ice_info->sess_state < PJ_ICE_STRANS_STATE_RUNNING) {
+ ice_complete = PJ_FALSE;
+ break;
+ }
+
+ /* Check if ICE needs to send reinvite */
+ if (!ice_need_reinv &&
+ ice_info->sess_state == PJ_ICE_STRANS_STATE_RUNNING &&
+ ice_info->role == PJ_ICE_SESS_ROLE_CONTROLLING)
+ {
+ pjsua_ice_config *cfg=&pjsua_var.acc[call->acc_id].cfg.ice_cfg;
+ if ((cfg->ice_always_update && !call->reinv_ice_sent) ||
+ pj_sockaddr_cmp(&tpinfo.sock_info.rtp_addr_name,
+ &call_med->rtp_addr))
+ {
+ ice_need_reinv = PJ_TRUE;
+ }
+ }
+ }
+
+ if (ice_complete && need_reinv)
+ *need_reinv = ice_need_reinv;
+
+ return ice_complete;
+}
+
+/* Check and send reinvite for lock codec and ICE update */
+static pj_status_t process_pending_reinvite(pjsua_call *call)
+{
+ const pj_str_t ST_UPDATE = {"UPDATE", 6};
+ pj_pool_t *pool = call->inv->pool_prov;
+ pjsip_inv_session *inv = call->inv;
+ pj_bool_t ice_need_reinv;
+ pj_bool_t ice_completed;
+ pj_bool_t need_lock_codec;
+ pj_bool_t rem_can_update;
+ pjmedia_sdp_session *new_offer;
+ pjsip_tx_data *tdata;
+ unsigned i;
+ pj_status_t status;
+
+ /* Verify if another SDP negotiation is in progress, e.g: session timer
+ * or another re-INVITE.
+ */
+ if (inv==NULL || inv->neg==NULL ||
+ pjmedia_sdp_neg_get_state(inv->neg)!=PJMEDIA_SDP_NEG_STATE_DONE)
+ {
+ return PJMEDIA_SDPNEG_EINSTATE;
+ }
+
+ /* Don't do this if call is disconnecting! */
+ if (inv->state > PJSIP_INV_STATE_CONFIRMED || inv->cause >= 200)
+ {
+ return PJ_EINVALIDOP;
+ }
+
+ /* Delay this when the SDP negotiation done in call state EARLY and
+ * remote does not support UPDATE method.
+ */
+ if (inv->state == PJSIP_INV_STATE_EARLY &&
+ pjsip_dlg_remote_has_cap(inv->dlg, PJSIP_H_ALLOW, NULL, &ST_UPDATE)!=
+ PJSIP_DIALOG_CAP_SUPPORTED)
+ {
+ call->reinv_pending = PJ_TRUE;
+ return PJ_EPENDING;
+ }
+
+ /* Check if ICE setup is complete and if it needs reinvite */
+ ice_completed = check_ice_complete(call, &ice_need_reinv);
+ if (!ice_completed)
+ return PJ_EPENDING;
+
+ /* Check if we need to lock codec */
+ need_lock_codec = check_lock_codec(call);
+
+ /* Check if reinvite is really needed */
+ if (!need_lock_codec && !ice_need_reinv)
+ return PJ_SUCCESS;
+
+
+ /* Okay! So we need to send re-INVITE/UPDATE */
+
+ /* Check if remote support UPDATE */
+ rem_can_update = pjsip_dlg_remote_has_cap(inv->dlg, PJSIP_H_ALLOW, NULL,
+ &ST_UPDATE) ==
+ PJSIP_DIALOG_CAP_SUPPORTED;
+
+ /* Logging stuff */
+ {
+ const char *ST_ICE_UPDATE = "ICE transport address after "
+ "ICE negotiation";
+ const char *ST_LOCK_CODEC = "media session to use only one codec";
+ PJ_LOG(4,(THIS_FILE, "Call %d sending %s for updating %s%s%s",
+ call->index,
+ (rem_can_update? "UPDATE" : "re-INVITE"),
+ (ice_need_reinv? ST_ICE_UPDATE : ST_LOCK_CODEC),
+ (ice_need_reinv && need_lock_codec? " and " : ""),
+ (ice_need_reinv && need_lock_codec? ST_LOCK_CODEC : "")
+ ));
+ }
+
+ /* Generate SDP re-offer */
+ status = pjsua_media_channel_create_sdp(call->index, pool, NULL,
+ &new_offer, NULL);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Unable to create local SDP", status);
+ return status;
+ }
+
+ /* Update the new offer so it contains only a codec. Note that
+ * SDP nego has removed unmatched codecs from the offer and the codec
+ * order in the offer has been matched to the answer, so we'll override
+ * the codecs in the just generated SDP with the ones from the active
+ * local SDP and leave just one codec for the next SDP re-offer.
+ */
+ if (need_lock_codec) {
+ const pjmedia_sdp_session *ref_sdp;
+
+ /* Get local active SDP as reference */
+ status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &ref_sdp);
+ if (status != PJ_SUCCESS)
+ return status;
+
+ /* Verify media count. Note that remote may add/remove media line
+ * in the answer. When answer has less media, it must have been
+ * handled by pjsua_media_channel_update() as disabled media.
+ * When answer has more media, it must have been ignored (treated
+ * as non-exist) anywhere. Local media count should not be updated
+ * at this point, as modifying media count operation (i.e: reinvite,
+ * update, vid_set_strm) is currently blocking, protected with
+ * dialog mutex, and eventually reset SDP nego state to LOCAL OFFER.
+ */
+ if (call->med_cnt != ref_sdp->media_count ||
+ ref_sdp->media_count != new_offer->media_count)
+ {
+ /* Anyway, just in case, let's just return error */
+ return PJMEDIA_SDPNEG_EINSTATE;
+ }
+
+ for (i = 0; i < call->med_cnt; ++i) {
+ unsigned j, codec_cnt = 0;
+ const pjmedia_sdp_media *ref_m = ref_sdp->media[i];
+ pjmedia_sdp_media *m = new_offer->media[i];
+ pjsua_call_media *call_med = &call->media[i];
+
+ /* Verify if media is deactivated */
+ if (call_med->state == PJSUA_CALL_MEDIA_NONE ||
+ call_med->state == PJSUA_CALL_MEDIA_ERROR ||
+ call_med->dir == PJMEDIA_DIR_NONE)
+ {
+ continue;
+ }
+
+ /* Reset formats */
+ m->desc.fmt_count = 0;
+ pjmedia_sdp_attr_remove_all(&m->attr_count, m->attr, "rtpmap");
+ pjmedia_sdp_attr_remove_all(&m->attr_count, m->attr, "fmtp");
+
+ /* Copy only the first format + any non-AV formats from
+ * the active local SDP.
+ */
+ for (j = 0; j < ref_m->desc.fmt_count; ++j) {
+ const pj_str_t *fmt = &ref_m->desc.fmt[j];
+
+ if (is_non_av_fmt(ref_m, fmt) || (++codec_cnt == 1)) {
+ pjmedia_sdp_attr *a;
+
+ m->desc.fmt[m->desc.fmt_count++] = *fmt;
+ a = pjmedia_sdp_attr_find2(ref_m->attr_count, ref_m->attr,
+ "rtpmap", fmt);
+ if (a) pjmedia_sdp_attr_add(&m->attr_count, m->attr, a);
+ a = pjmedia_sdp_attr_find2(ref_m->attr_count, ref_m->attr,
+ "fmtp", fmt);
+ if (a) pjmedia_sdp_attr_add(&m->attr_count, m->attr, a);
+ }
+ }
+ }
+ }
+
+ /* Put back original direction and "c=0.0.0.0" line */
+ {
+ const pjmedia_sdp_session *cur_sdp;
+
+ /* Get local active SDP */
+ status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &cur_sdp);
+ if (status != PJ_SUCCESS)
+ return status;
+
+ /* Make sure media count has not been changed */
+ if (call->med_cnt != cur_sdp->media_count)
+ return PJMEDIA_SDPNEG_EINSTATE;
+
+ for (i = 0; i < call->med_cnt; ++i) {
+ const pjmedia_sdp_media *m = cur_sdp->media[i];
+ pjmedia_sdp_media *new_m = new_offer->media[i];
+ pjsua_call_media *call_med = &call->media[i];
+ pjmedia_sdp_attr *a = NULL;
+
+ /* Update direction to the current dir */
+ pjmedia_sdp_media_remove_all_attr(new_m, "sendrecv");
+ pjmedia_sdp_media_remove_all_attr(new_m, "sendonly");
+ pjmedia_sdp_media_remove_all_attr(new_m, "recvonly");
+ pjmedia_sdp_media_remove_all_attr(new_m, "inactive");
+
+ if (call_med->dir == PJMEDIA_DIR_ENCODING_DECODING) {
+ a = pjmedia_sdp_attr_create(pool, "sendrecv", NULL);
+ } else if (call_med->dir == PJMEDIA_DIR_ENCODING) {
+ a = pjmedia_sdp_attr_create(pool, "sendonly", NULL);
+ } else if (call_med->dir == PJMEDIA_DIR_DECODING) {
+ a = pjmedia_sdp_attr_create(pool, "recvonly", NULL);
+ } else {
+ const pjmedia_sdp_conn *conn;
+ a = pjmedia_sdp_attr_create(pool, "inactive", NULL);
+
+ /* Also check if the original c= line address is zero */
+ conn = m->conn;
+ if (!conn)
+ conn = cur_sdp->conn;
+ if (pj_strcmp2(&conn->addr, "0.0.0.0")==0 ||
+ pj_strcmp2(&conn->addr, "0")==0)
+ {
+ if (!new_m->conn) {
+ new_m->conn = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_conn);
+ }
+
+ if (pj_strcmp2(&new_m->conn->addr, "0.0.0.0")) {
+ new_m->conn->net_type = pj_str("IN");
+ new_m->conn->addr_type = pj_str("IP4");
+ new_m->conn->addr = pj_str("0.0.0.0");
+ }
+ }
+ }
+
+ pj_assert(a);
+ pjmedia_sdp_media_add_attr(new_m, a);
+ }
+ }
+
+
+ if (rem_can_update) {
+ status = pjsip_inv_update(inv, NULL, new_offer, &tdata);
+ } else {
+ status = pjsip_inv_reinvite(inv, NULL, new_offer, &tdata);
+ }
+
+ if (status==PJ_EINVALIDOP &&
+ ++call->lock_codec.retry_cnt < LOCK_CODEC_MAX_RETRY)
+ {
+ /* Ups, let's reschedule again */
+ pjsua_call_schedule_reinvite_check(call, LOCK_CODEC_RETRY_INTERVAL);
+ return PJ_SUCCESS;
+ } else if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Error creating UPDATE/re-INVITE",
+ status);
+ return status;
+ }
+
+ /* Send the UPDATE/re-INVITE request */
+ status = pjsip_inv_send_msg(inv, tdata);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Error sending UPDATE/re-INVITE",
+ status);
+ return status;
+ }
+
+ /* Update flags */
+ if (ice_need_reinv)
+ call->reinv_ice_sent = PJ_TRUE;
+ if (need_lock_codec)
+ ++call->lock_codec.retry_cnt;
+
+ return PJ_SUCCESS;
+}
+
+
+/*
+ * This callback receives notification from invite session when the
+ * session state has changed.
+ */
+static void pjsua_call_on_state_changed(pjsip_inv_session *inv,
+ pjsip_event *e)
+{
+ pjsua_call *call;
+
+ pj_log_push_indent();
+
+ call = (pjsua_call*) inv->dlg->mod_data[pjsua_var.mod.id];
+
+ if (!call) {
+ pj_log_pop_indent();
+ return;
+ }
+
+
+ /* Get call times */
+ switch (inv->state) {
+ case PJSIP_INV_STATE_EARLY:
+ case PJSIP_INV_STATE_CONNECTING:
+ if (call->res_time.sec == 0)
+ pj_gettimeofday(&call->res_time);
+ call->last_code = (pjsip_status_code)
+ e->body.tsx_state.tsx->status_code;
+ pj_strncpy(&call->last_text,
+ &e->body.tsx_state.tsx->status_text,
+ sizeof(call->last_text_buf_));
+ break;
+ case PJSIP_INV_STATE_CONFIRMED:
+ pj_gettimeofday(&call->conn_time);
+
+ /* See if auto reinvite was pended as media update was done in the
+ * EARLY state and remote does not support UPDATE.
+ */
+ if (call->reinv_pending) {
+ call->reinv_pending = PJ_FALSE;
+ pjsua_call_schedule_reinvite_check(call, 0);
+ }
+ break;
+ case PJSIP_INV_STATE_DISCONNECTED:
+ pj_gettimeofday(&call->dis_time);
+ if (call->res_time.sec == 0)
+ pj_gettimeofday(&call->res_time);
+ if (e->type == PJSIP_EVENT_TSX_STATE &&
+ e->body.tsx_state.tsx->status_code > call->last_code)
+ {
+ call->last_code = (pjsip_status_code)
+ e->body.tsx_state.tsx->status_code;
+ pj_strncpy(&call->last_text,
+ &e->body.tsx_state.tsx->status_text,
+ sizeof(call->last_text_buf_));
+ } else {
+ call->last_code = PJSIP_SC_REQUEST_TERMINATED;
+ pj_strncpy(&call->last_text,
+ pjsip_get_status_text(call->last_code),
+ sizeof(call->last_text_buf_));
+ }
+
+ /* Stop reinvite timer, if it is active */
+ if (call->reinv_timer.id) {
+ pjsua_cancel_timer(&call->reinv_timer);
+ call->reinv_timer.id = PJ_FALSE;
+ }
+ break;
+ default:
+ call->last_code = (pjsip_status_code)
+ e->body.tsx_state.tsx->status_code;
+ pj_strncpy(&call->last_text,
+ &e->body.tsx_state.tsx->status_text,
+ sizeof(call->last_text_buf_));
+ break;
+ }
+
+ /* If this is an outgoing INVITE that was created because of
+ * REFER/transfer, send NOTIFY to transferer.
+ */
+ if (call->xfer_sub && e->type==PJSIP_EVENT_TSX_STATE) {
+ int st_code = -1;
+ pjsip_evsub_state ev_state = PJSIP_EVSUB_STATE_ACTIVE;
+
+
+ switch (call->inv->state) {
+ case PJSIP_INV_STATE_NULL:
+ case PJSIP_INV_STATE_CALLING:
+ /* Do nothing */
+ break;
+
+ case PJSIP_INV_STATE_EARLY:
+ case PJSIP_INV_STATE_CONNECTING:
+ st_code = e->body.tsx_state.tsx->status_code;
+ if (call->inv->state == PJSIP_INV_STATE_CONNECTING)
+ ev_state = PJSIP_EVSUB_STATE_TERMINATED;
+ else
+ ev_state = PJSIP_EVSUB_STATE_ACTIVE;
+ break;
+
+ case PJSIP_INV_STATE_CONFIRMED:
+#if 0
+/* We don't need this, as we've terminated the subscription in
+ * CONNECTING state.
+ */
+ /* When state is confirmed, send the final 200/OK and terminate
+ * subscription.
+ */
+ st_code = e->body.tsx_state.tsx->status_code;
+ ev_state = PJSIP_EVSUB_STATE_TERMINATED;
+#endif
+ break;
+
+ case PJSIP_INV_STATE_DISCONNECTED:
+ st_code = e->body.tsx_state.tsx->status_code;
+ ev_state = PJSIP_EVSUB_STATE_TERMINATED;
+ break;
+
+ case PJSIP_INV_STATE_INCOMING:
+ /* Nothing to do. Just to keep gcc from complaining about
+ * unused enums.
+ */
+ break;
+ }
+
+ if (st_code != -1) {
+ pjsip_tx_data *tdata;
+ pj_status_t status;
+
+ status = pjsip_xfer_notify( call->xfer_sub,
+ ev_state, st_code,
+ NULL, &tdata);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Unable to create NOTIFY", status);
+ } else {
+ status = pjsip_xfer_send_request(call->xfer_sub, tdata);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Unable to send NOTIFY", status);
+ }
+ }
+ }
+ }
+
+ /* Ticket #1627: Invoke on_call_tsx_state() when call is disconnected. */
+ if (inv->state == PJSIP_INV_STATE_DISCONNECTED &&
+ e->type == PJSIP_EVENT_TSX_STATE &&
+ call->inv &&
+ pjsua_var.ua_cfg.cb.on_call_tsx_state)
+ {
+ (*pjsua_var.ua_cfg.cb.on_call_tsx_state)(call->index,
+ e->body.tsx_state.tsx, e);
+ }
+
+ if (pjsua_var.ua_cfg.cb.on_call_state)
+ (*pjsua_var.ua_cfg.cb.on_call_state)(call->index, e);
+
+ /* 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. */
+ pjsua_call_schedule_reinvite_check(call, 0);
+
+ /* Call application callback, if any */
+ if (pjsua_var.ua_cfg.cb.on_call_media_state)
+ pjsua_var.ua_cfg.cb.on_call_media_state(call->index);
+
+on_return:
+ pj_log_pop_indent();
+}
+
+
+/* Modify SDP for call hold. */
+static pj_status_t modify_sdp_of_call_hold(pjsua_call *call,
+ pj_pool_t *pool,
+ pjmedia_sdp_session *sdp,
+ pj_bool_t as_answerer)
+{
+ unsigned mi;
+
+ /* Call-hold is done by set the media direction to 'sendonly'
+ * (PJMEDIA_DIR_ENCODING), except when current media direction is
+ * 'inactive' (PJMEDIA_DIR_NONE).
+ * (See RFC 3264 Section 8.4 and RFC 4317 Section 3.1)
+ */
+ /* http://trac.pjsip.org/repos/ticket/880
+ if (call->dir != PJMEDIA_DIR_ENCODING) {
+ */
+ /* https://trac.pjsip.org/repos/ticket/1142:
+ * configuration to use c=0.0.0.0 for call hold.
+ */
+
+ for (mi=0; 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");
+
+ /* When as answerer, just simply set dir to "sendonly", note that
+ * if the offer uses "sendonly" or "inactive", the SDP negotiator
+ * will change our answer dir to "inactive".
+ */
+ if (as_answerer || (call->media[mi].dir & PJMEDIA_DIR_ENCODING)) {
+ /* Add sendonly attribute */
+ attr = pjmedia_sdp_attr_create(pool, "sendonly", NULL);
+ pjmedia_sdp_media_add_attr(m, attr);
+ } else {
+ /* Add inactive attribute */
+ attr = pjmedia_sdp_attr_create(pool, "inactive", NULL);
+ pjmedia_sdp_media_add_attr(m, attr);
+ }
+ }
+ }
+
+ return PJ_SUCCESS;
+}
+
+/* Create SDP for call hold. */
+static pj_status_t create_sdp_of_call_hold(pjsua_call *call,
+ pjmedia_sdp_session **p_sdp)
+{
+ pj_status_t status;
+ pj_pool_t *pool;
+ pjmedia_sdp_session *sdp;
+
+ /* Use call's provisional pool */
+ pool = call->inv->pool_prov;
+
+ /* Create new offer */
+ status = pjsua_media_channel_create_sdp(call->index, pool, NULL, &sdp,
+ NULL);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Unable to create local SDP", status);
+ return status;
+ }
+
+ status = modify_sdp_of_call_hold(call, pool, sdp, PJ_FALSE);
+ if (status != PJ_SUCCESS)
+ return status;
+
+ *p_sdp = sdp;
+
+ return PJ_SUCCESS;
+}
+
+/*
+ * Called when session received new offer.
+ */
+static void pjsua_call_on_rx_offer(pjsip_inv_session *inv,
+ 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, PJ_TRUE);
+ }
+
+ status = pjsip_inv_set_sdp_answer(call->inv, answer);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Unable to set answer", status);
+ goto on_return;
+ }
+
+on_return:
+ pj_log_pop_indent();
+}
+
+
+/*
+ * Called 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) {
+ /* Call has been disconnected. */
+ goto on_return;
+ }
+
+ /* https://trac.pjsip.org/repos/ticket/1452:
+ * If a request is retried due to 401/407 challenge, don't process the
+ * transaction first but wait until we've retried it.
+ */
+ if (tsx->role == PJSIP_ROLE_UAC &&
+ (tsx->status_code==401 || tsx->status_code==407) &&
+ tsx->last_tx && tsx->last_tx->auth_retry)
+ {
+ goto on_return;
+ }
+
+ /* Notify application callback first */
+ if (pjsua_var.ua_cfg.cb.on_call_tsx_state) {
+ (*pjsua_var.ua_cfg.cb.on_call_tsx_state)(call->index, tsx, e);
+ }
+
+ if (tsx->role==PJSIP_ROLE_UAS &&
+ tsx->state==PJSIP_TSX_STATE_TRYING &&
+ pjsip_method_cmp(&tsx->method, pjsip_get_refer_method())==0)
+ {
+ /*
+ * Incoming REFER request.
+ */
+ on_call_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_UAC &&
+ (call->opt.flag & PJSUA_CALL_UNHOLD) &&
+ tsx->state >= PJSIP_TSX_STATE_COMPLETED)
+ {
+ /* Monitor the status of call unhold request */
+ if (tsx->status_code/100 != 2 &&
+ (tsx->status_code!=401 && tsx->status_code!=407))
+ {
+ /* Call unhold failed */
+ call->opt.flag &= ~PJSUA_CALL_UNHOLD;
+ call->local_hold = PJ_TRUE;
+ PJ_LOG(3,(THIS_FILE, "Error releasing hold on call %d (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;
+}
+
diff --git a/jni/pjproject-android/.svn/pristine/73/73b1c9996dae322c9a03554ba8cf9ddbefa1c29a.svn-base b/jni/pjproject-android/.svn/pristine/73/73b1c9996dae322c9a03554ba8cf9ddbefa1c29a.svn-base
new file mode 100644
index 0000000..f168437
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/73/73b1c9996dae322c9a03554ba8cf9ddbefa1c29a.svn-base
@@ -0,0 +1,20 @@
+<?xml version="1.0" encoding="Windows-1252"?>
+<VisualStudioPropertySheet
+ ProjectType="Visual C++"
+ Version="8.00"
+ Name="pjproject-vs8-win64-common-defaults"
+ >
+ <Tool
+ Name="VCCLCompilerTool"
+ PreprocessorDefinitions="WIN64;PJ_WIN64=1;PJ_M_X86_64=1;"
+ />
+ <Tool
+ Name="VCLinkerTool"
+ SubSystem="1"
+ TargetMachine="17"
+ />
+ <UserMacro
+ Name="TargetCPU"
+ Value="x86_64"
+ />
+</VisualStudioPropertySheet>
diff --git a/jni/pjproject-android/.svn/pristine/73/73fbab59bc49b0ec72d3560cd09c3a90a25c9a2a.svn-base b/jni/pjproject-android/.svn/pristine/73/73fbab59bc49b0ec72d3560cd09c3a90a25c9a2a.svn-base
new file mode 100644
index 0000000..0290fea
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/73/73fbab59bc49b0ec72d3560cd09c3a90a25c9a2a.svn-base
@@ -0,0 +1,33 @@
+#
+# OS specific configuration for Win32 OS target.
+#
+
+#
+# PJLIB_OBJS specified here are object files to be included in PJLIB
+# (the library) for this specific operating system. Object files common
+# to all operating systems should go in Makefile instead.
+#
+export PJLIB_OBJS += addr_resolv_sock.o guid_win32.o \
+ log_writer_stdout.o os_core_win32.o \
+ os_error_win32.o os_time_bsd.o os_timestamp_common.o \
+ os_timestamp_win32.o \
+ pool_policy_malloc.o sock_bsd.o sock_select.o
+
+#export PJLIB_OBJS += ioqueue_winnt.o
+export PJLIB_OBJS += ioqueue_select.o
+
+export PJLIB_OBJS += file_io_win32.o file_access_win32.o
+#export PJLIB_OBJS += file_io_ansi.o
+
+#
+# TEST_OBJS are operating system specific object files to be included in
+# the test application.
+#
+export TEST_OBJS += main.o
+
+#
+# TARGETS are make targets in the Makefile, to be executed for this given
+# operating system.
+#
+export TARGETS = pjlib pjlib-test
+