* #39226: Switch back to pjsip rev 4710
Rev 4716 introduces errors when building for android (miltiple definitions)
diff --git a/jni/pjproject-android/.svn/pristine/5e/5ede33e106763f923e018c480baf530aef25cab1.svn-base b/jni/pjproject-android/.svn/pristine/5e/5ede33e106763f923e018c480baf530aef25cab1.svn-base
new file mode 100644
index 0000000..ce7722c
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/5e/5ede33e106763f923e018c480baf530aef25cab1.svn-base
@@ -0,0 +1,3061 @@
+/* $Id$ */
+/*
+ * 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_media.c"
+
+#define DEFAULT_RTP_PORT 4000
+
+#ifndef PJSUA_REQUIRE_CONSECUTIVE_RTCP_PORT
+# define PJSUA_REQUIRE_CONSECUTIVE_RTCP_PORT 0
+#endif
+
+static void pjsua_media_config_dup(pj_pool_t *pool,
+ pjsua_media_config *dst,
+ const pjsua_media_config *src)
+{
+ pj_memcpy(dst, src, sizeof(*src));
+ pj_strdup(pool, &dst->turn_server, &src->turn_server);
+ pj_stun_auth_cred_dup(pool, &dst->turn_auth_cred, &src->turn_auth_cred);
+}
+
+
+/**
+ * Init media subsystems.
+ */
+pj_status_t pjsua_media_subsys_init(const pjsua_media_config *cfg)
+{
+ pj_status_t status;
+
+ pj_log_push_indent();
+
+ /* Specify which audio device settings are save-able */
+ pjsua_var.aud_svmask = 0xFFFFFFFF;
+ /* These are not-settable */
+ pjsua_var.aud_svmask &= ~(PJMEDIA_AUD_DEV_CAP_EXT_FORMAT |
+ PJMEDIA_AUD_DEV_CAP_INPUT_SIGNAL_METER |
+ PJMEDIA_AUD_DEV_CAP_OUTPUT_SIGNAL_METER);
+ /* EC settings use different API */
+ pjsua_var.aud_svmask &= ~(PJMEDIA_AUD_DEV_CAP_EC |
+ PJMEDIA_AUD_DEV_CAP_EC_TAIL);
+
+ /* Copy configuration */
+ pjsua_media_config_dup(pjsua_var.pool, &pjsua_var.media_cfg, cfg);
+
+ /* Normalize configuration */
+ if (pjsua_var.media_cfg.snd_clock_rate == 0) {
+ pjsua_var.media_cfg.snd_clock_rate = pjsua_var.media_cfg.clock_rate;
+ }
+
+ if (pjsua_var.media_cfg.has_ioqueue &&
+ pjsua_var.media_cfg.thread_cnt == 0)
+ {
+ pjsua_var.media_cfg.thread_cnt = 1;
+ }
+
+ if (pjsua_var.media_cfg.max_media_ports < pjsua_var.ua_cfg.max_calls) {
+ pjsua_var.media_cfg.max_media_ports = pjsua_var.ua_cfg.max_calls + 2;
+ }
+
+ /* Create media endpoint. */
+ status = pjmedia_endpt_create(&pjsua_var.cp.factory,
+ pjsua_var.media_cfg.has_ioqueue? NULL :
+ pjsip_endpt_get_ioqueue(pjsua_var.endpt),
+ pjsua_var.media_cfg.thread_cnt,
+ &pjsua_var.med_endpt);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE,
+ "Media stack initialization has returned error",
+ status);
+ goto on_error;
+ }
+
+ status = pjsua_aud_subsys_init();
+ if (status != PJ_SUCCESS)
+ goto on_error;
+
+#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
+ /* Initialize SRTP library (ticket #788). */
+ status = pjmedia_srtp_init_lib(pjsua_var.med_endpt);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Error initializing SRTP library",
+ status);
+ goto on_error;
+ }
+#endif
+
+ /* Video */
+#if PJMEDIA_HAS_VIDEO
+ status = pjsua_vid_subsys_init();
+ if (status != PJ_SUCCESS)
+ goto on_error;
+#endif
+
+ pj_log_pop_indent();
+ return PJ_SUCCESS;
+
+on_error:
+ pj_log_pop_indent();
+ return status;
+}
+
+/*
+ * Start pjsua media subsystem.
+ */
+pj_status_t pjsua_media_subsys_start(void)
+{
+ pj_status_t status;
+
+ pj_log_push_indent();
+
+#if DISABLED_FOR_TICKET_1185
+ /* Create media for calls, if none is specified */
+ if (pjsua_var.calls[0].media[0].tp == NULL) {
+ pjsua_transport_config transport_cfg;
+
+ /* Create default transport config */
+ pjsua_transport_config_default(&transport_cfg);
+ transport_cfg.port = DEFAULT_RTP_PORT;
+
+ status = pjsua_media_transports_create(&transport_cfg);
+ if (status != PJ_SUCCESS) {
+ pj_log_pop_indent();
+ return status;
+ }
+ }
+#endif
+
+ /* Audio */
+ status = pjsua_aud_subsys_start();
+ if (status != PJ_SUCCESS) {
+ pj_log_pop_indent();
+ return status;
+ }
+
+ /* Video */
+#if PJMEDIA_HAS_VIDEO
+ status = pjsua_vid_subsys_start();
+ if (status != PJ_SUCCESS) {
+ pjsua_aud_subsys_destroy();
+ pj_log_pop_indent();
+ return status;
+ }
+#endif
+
+ /* Perform NAT detection */
+ if (pjsua_var.ua_cfg.stun_srv_cnt) {
+ status = pjsua_detect_nat_type();
+ if (status != PJ_SUCCESS) {
+ PJ_PERROR(1,(THIS_FILE, status, "NAT type detection failed"));
+ }
+ }
+
+ pj_log_pop_indent();
+ return PJ_SUCCESS;
+}
+
+
+/*
+ * Destroy pjsua media subsystem.
+ */
+pj_status_t pjsua_media_subsys_destroy(unsigned flags)
+{
+ PJ_UNUSED_ARG(flags);
+
+ PJ_LOG(4,(THIS_FILE, "Shutting down media.."));
+ pj_log_push_indent();
+
+ if (pjsua_var.med_endpt) {
+ /* Wait for media endpoint's worker threads to quit. */
+ pjmedia_endpt_stop_threads(pjsua_var.med_endpt);
+
+ pjsua_aud_subsys_destroy();
+ }
+
+#if 0
+ // This part has been moved out to pjsua_destroy() (see also #1717).
+ /* Close media transports */
+ for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) {
+ /* TODO: check if we're not allowed to send to network in the
+ * "flags", and if so do not do TURN allocation...
+ */
+ PJ_UNUSED_ARG(flags);
+ pjsua_media_channel_deinit(i);
+ }
+#endif
+
+ /* Destroy media endpoint. */
+ if (pjsua_var.med_endpt) {
+
+# if PJMEDIA_HAS_VIDEO
+ pjsua_vid_subsys_destroy();
+# endif
+
+ pjmedia_endpt_destroy(pjsua_var.med_endpt);
+ pjsua_var.med_endpt = NULL;
+
+ /* Deinitialize sound subsystem */
+ // Not necessary, as pjmedia_snd_deinit() should have been called
+ // in pjmedia_endpt_destroy().
+ //pjmedia_snd_deinit();
+ }
+
+ pj_log_pop_indent();
+
+ return PJ_SUCCESS;
+}
+
+/*
+ * Create RTP and RTCP socket pair, and possibly resolve their public
+ * address via STUN.
+ */
+static pj_status_t create_rtp_rtcp_sock(pjsua_call_media *call_med,
+ const pjsua_transport_config *cfg,
+ pjmedia_sock_info *skinfo)
+{
+ enum {
+ RTP_RETRY = 100
+ };
+ int i;
+ pj_bool_t use_ipv6;
+ int af;
+ pj_sockaddr bound_addr;
+ pj_sockaddr mapped_addr[2];
+ pj_status_t status = PJ_SUCCESS;
+ char addr_buf[PJ_INET6_ADDRSTRLEN+10];
+ pjsua_acc *acc = &pjsua_var.acc[call_med->call->acc_id];
+ pj_sock_t sock[2];
+
+ use_ipv6 = (acc->cfg.ipv6_media_use != PJSUA_IPV6_DISABLED);
+ af = use_ipv6 ? pj_AF_INET6() : pj_AF_INET();
+
+ /* Make sure STUN server resolution has completed */
+ if (!use_ipv6 && pjsua_sip_acc_is_using_stun(call_med->call->acc_id)) {
+ status = resolve_stun_server(PJ_TRUE);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Error resolving STUN server", status);
+ return status;
+ }
+ }
+
+ if (acc->next_rtp_port == 0)
+ acc->next_rtp_port = (pj_uint16_t)cfg->port;
+
+ if (acc->next_rtp_port == 0)
+ acc->next_rtp_port = (pj_uint16_t)DEFAULT_RTP_PORT;
+
+ for (i=0; i<2; ++i)
+ sock[i] = PJ_INVALID_SOCKET;
+
+ pj_sockaddr_init(af, &bound_addr, NULL, 0);
+ if (cfg->bound_addr.slen) {
+ status = pj_sockaddr_set_str_addr(af, &bound_addr, &cfg->bound_addr);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Unable to resolve transport bind address",
+ status);
+ return status;
+ }
+ }
+
+ /* Loop retry to bind RTP and RTCP sockets. */
+ for (i=0; i<RTP_RETRY; ++i, acc->next_rtp_port += 2) {
+
+ if (cfg->port > 0 && cfg->port_range > 0 &&
+ (acc->next_rtp_port > cfg->port + cfg->port_range ||
+ acc->next_rtp_port < cfg->port))
+ {
+ acc->next_rtp_port = (pj_uint16_t)cfg->port;
+ }
+
+ /* Create RTP socket. */
+ status = pj_sock_socket(af, pj_SOCK_DGRAM(), 0, &sock[0]);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "socket() error", status);
+ return status;
+ }
+
+ /* Apply QoS to RTP socket, if specified */
+ status = pj_sock_apply_qos2(sock[0], cfg->qos_type,
+ &cfg->qos_params,
+ 2, THIS_FILE, "RTP socket");
+
+ /* Bind RTP socket */
+ pj_sockaddr_set_port(&bound_addr, acc->next_rtp_port);
+ status=pj_sock_bind(sock[0], &bound_addr,
+ pj_sockaddr_get_len(&bound_addr));
+ if (status != PJ_SUCCESS) {
+ pj_sock_close(sock[0]);
+ sock[0] = PJ_INVALID_SOCKET;
+ continue;
+ }
+
+ /* Create RTCP socket. */
+ status = pj_sock_socket(af, pj_SOCK_DGRAM(), 0, &sock[1]);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "socket() error", status);
+ pj_sock_close(sock[0]);
+ return status;
+ }
+
+ /* Apply QoS to RTCP socket, if specified */
+ status = pj_sock_apply_qos2(sock[1], cfg->qos_type,
+ &cfg->qos_params,
+ 2, THIS_FILE, "RTCP socket");
+
+ /* Bind RTCP socket */
+ pj_sockaddr_set_port(&bound_addr, (pj_uint16_t)(acc->next_rtp_port+1));
+ status=pj_sock_bind(sock[1], &bound_addr,
+ pj_sockaddr_get_len(&bound_addr));
+ if (status != PJ_SUCCESS) {
+ pj_sock_close(sock[0]);
+ sock[0] = PJ_INVALID_SOCKET;
+
+ pj_sock_close(sock[1]);
+ sock[1] = PJ_INVALID_SOCKET;
+ continue;
+ }
+
+ /*
+ * If we're configured to use STUN, then find out the mapped address,
+ * and make sure that the mapped RTCP port is adjacent with the RTP.
+ */
+ if (!use_ipv6 && pjsua_sip_acc_is_using_stun(call_med->call->acc_id) &&
+ pjsua_var.stun_srv.addr.sa_family != 0)
+ {
+ char ip_addr[32];
+ pj_str_t stun_srv;
+ pj_sockaddr_in resolved_addr[2];
+ pjstun_setting stun_opt;
+
+ pj_ansi_strcpy(ip_addr,
+ pj_inet_ntoa(pjsua_var.stun_srv.ipv4.sin_addr));
+ stun_srv = pj_str(ip_addr);
+
+ pj_bzero(&stun_opt, sizeof(stun_opt));
+ stun_opt.use_stun2 = pjsua_var.ua_cfg.stun_map_use_stun2;
+ stun_opt.srv1 = stun_opt.srv2 = stun_srv;
+ stun_opt.port1 = stun_opt.port2 =
+ pj_ntohs(pjsua_var.stun_srv.ipv4.sin_port);
+ status=pjstun_get_mapped_addr2(&pjsua_var.cp.factory, &stun_opt,
+ 2, sock, resolved_addr);
+#if defined(PJ_IPHONE_OS_HAS_MULTITASKING_SUPPORT) && \
+ PJ_IPHONE_OS_HAS_MULTITASKING_SUPPORT!=0
+ /* Handle EPIPE (Broken Pipe) error, which happens on UDP socket
+ * after app wakes up from suspended state. In this case, simply
+ * just retry.
+ * P.S.: The magic status is PJ_STATUS_FROM_OS(EPIPE)
+ */
+ if (status == 120032) {
+ PJ_LOG(4,(THIS_FILE, "Got EPIPE error, retrying.."));
+ pj_sock_close(sock[0]);
+ sock[0] = PJ_INVALID_SOCKET;
+
+ pj_sock_close(sock[1]);
+ sock[1] = PJ_INVALID_SOCKET;
+
+ continue;
+ }
+ else
+#endif
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "STUN resolve error", status);
+ goto on_error;
+ }
+
+ pj_sockaddr_cp(&mapped_addr[0], &resolved_addr[0]);
+ pj_sockaddr_cp(&mapped_addr[1], &resolved_addr[1]);
+
+#if PJSUA_REQUIRE_CONSECUTIVE_RTCP_PORT
+ if (pj_sockaddr_get_port(&mapped_addr[1]) ==
+ pj_sockaddr_get_port(&mapped_addr[0])+1)
+ {
+ /* Success! */
+ break;
+ }
+
+ pj_sock_close(sock[0]);
+ sock[0] = PJ_INVALID_SOCKET;
+
+ pj_sock_close(sock[1]);
+ sock[1] = PJ_INVALID_SOCKET;
+#else
+ if (pj_sockaddr_get_port(&mapped_addr[1]) !=
+ pj_sockaddr_get_port(&mapped_addr[0])+1)
+ {
+ PJ_LOG(4,(THIS_FILE,
+ "Note: STUN mapped RTCP port %d is not adjacent"
+ " to RTP port %d",
+ pj_sockaddr_get_port(&mapped_addr[1]),
+ pj_sockaddr_get_port(&mapped_addr[0])));
+ }
+ /* Success! */
+ break;
+#endif
+
+ } else if (cfg->public_addr.slen) {
+
+ status = pj_sockaddr_init(af, &mapped_addr[0], &cfg->public_addr,
+ (pj_uint16_t)acc->next_rtp_port);
+ if (status != PJ_SUCCESS)
+ goto on_error;
+
+ status = pj_sockaddr_init(af, &mapped_addr[1], &cfg->public_addr,
+ (pj_uint16_t)(acc->next_rtp_port+1));
+ if (status != PJ_SUCCESS)
+ goto on_error;
+
+ break;
+
+ } else {
+ if (acc->cfg.allow_sdp_nat_rewrite && acc->reg_mapped_addr.slen) {
+ pj_status_t status;
+
+ /* Take the address from mapped addr as seen by registrar */
+ status = pj_sockaddr_set_str_addr(af, &bound_addr,
+ &acc->reg_mapped_addr);
+ if (status != PJ_SUCCESS) {
+ /* just leave bound_addr with whatever it was
+ pj_bzero(pj_sockaddr_get_addr(&bound_addr),
+ pj_sockaddr_get_addr_len(&bound_addr));
+ */
+ }
+ }
+
+ if (!pj_sockaddr_has_addr(&bound_addr)) {
+ pj_sockaddr addr;
+
+ /* Get local IP address. */
+ status = pj_gethostip(af, &addr);
+ if (status != PJ_SUCCESS)
+ goto on_error;
+
+ pj_sockaddr_copy_addr(&bound_addr, &addr);
+ }
+
+ for (i=0; i<2; ++i) {
+ pj_sockaddr_init(af, &mapped_addr[i], NULL, 0);
+ pj_sockaddr_copy_addr(&mapped_addr[i], &bound_addr);
+ pj_sockaddr_set_port(&mapped_addr[i],
+ (pj_uint16_t)(acc->next_rtp_port+i));
+ }
+
+ break;
+ }
+ }
+
+ if (sock[0] == PJ_INVALID_SOCKET) {
+ PJ_LOG(1,(THIS_FILE,
+ "Unable to find appropriate RTP/RTCP ports combination"));
+ goto on_error;
+ }
+
+
+ skinfo->rtp_sock = sock[0];
+ pj_sockaddr_cp(&skinfo->rtp_addr_name, &mapped_addr[0]);
+
+ skinfo->rtcp_sock = sock[1];
+ pj_sockaddr_cp(&skinfo->rtcp_addr_name, &mapped_addr[1]);
+
+ PJ_LOG(4,(THIS_FILE, "RTP socket reachable at %s",
+ pj_sockaddr_print(&skinfo->rtp_addr_name, addr_buf,
+ sizeof(addr_buf), 3)));
+ PJ_LOG(4,(THIS_FILE, "RTCP socket reachable at %s",
+ pj_sockaddr_print(&skinfo->rtcp_addr_name, addr_buf,
+ sizeof(addr_buf), 3)));
+
+ acc->next_rtp_port += 2;
+ return PJ_SUCCESS;
+
+on_error:
+ for (i=0; i<2; ++i) {
+ if (sock[i] != PJ_INVALID_SOCKET)
+ pj_sock_close(sock[i]);
+ }
+ return status;
+}
+
+/* Create normal UDP media transports */
+static pj_status_t create_udp_media_transport(const pjsua_transport_config *cfg,
+ pjsua_call_media *call_med)
+{
+ pjmedia_sock_info skinfo;
+ pj_status_t status;
+
+ status = create_rtp_rtcp_sock(call_med, cfg, &skinfo);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Unable to create RTP/RTCP socket",
+ status);
+ goto on_error;
+ }
+
+ status = pjmedia_transport_udp_attach(pjsua_var.med_endpt, NULL,
+ &skinfo, 0, &call_med->tp);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Unable to create media transport",
+ status);
+ goto on_error;
+ }
+
+ pjmedia_transport_simulate_lost(call_med->tp, PJMEDIA_DIR_ENCODING,
+ pjsua_var.media_cfg.tx_drop_pct);
+
+ pjmedia_transport_simulate_lost(call_med->tp, PJMEDIA_DIR_DECODING,
+ pjsua_var.media_cfg.rx_drop_pct);
+
+ call_med->tp_ready = PJ_SUCCESS;
+
+ return PJ_SUCCESS;
+
+on_error:
+ if (call_med->tp)
+ pjmedia_transport_close(call_med->tp);
+
+ return status;
+}
+
+#if DISABLED_FOR_TICKET_1185
+/* Create normal UDP media transports */
+static pj_status_t create_udp_media_transports(pjsua_transport_config *cfg)
+{
+ unsigned i;
+ pj_status_t status;
+
+ for (i=0; i < pjsua_var.ua_cfg.max_calls; ++i) {
+ pjsua_call *call = &pjsua_var.calls[i];
+ unsigned strm_idx;
+
+ for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) {
+ pjsua_call_media *call_med = &call->media[strm_idx];
+
+ status = create_udp_media_transport(cfg, &call_med->tp);
+ if (status != PJ_SUCCESS)
+ goto on_error;
+ }
+ }
+
+ return PJ_SUCCESS;
+
+on_error:
+ for (i=0; i < pjsua_var.ua_cfg.max_calls; ++i) {
+ pjsua_call *call = &pjsua_var.calls[i];
+ unsigned strm_idx;
+
+ for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) {
+ pjsua_call_media *call_med = &call->media[strm_idx];
+
+ if (call_med->tp) {
+ pjmedia_transport_close(call_med->tp);
+ call_med->tp = NULL;
+ }
+ }
+ }
+ return status;
+}
+#endif
+
+/* Deferred callback to notify ICE init complete */
+static void ice_init_complete_cb(void *user_data)
+{
+ pjsua_call_media *call_med = (pjsua_call_media*)user_data;
+
+ if (call_med->call == NULL)
+ return;
+
+ /* No need to acquire_call() if we only change the tp_ready flag
+ * (i.e. transport is being created synchronously). Otherwise
+ * calling acquire_call() here may cause deadlock. See
+ * https://trac.pjsip.org/repos/ticket/1578
+ */
+ call_med->tp_ready = call_med->tp_result;
+
+ if (call_med->med_create_cb) {
+ pjsua_call *call = NULL;
+ pjsip_dialog *dlg = NULL;
+
+ if (acquire_call("ice_init_complete_cb", call_med->call->index,
+ &call, &dlg) != PJ_SUCCESS)
+ {
+ /* Call have been terminated */
+ return;
+ }
+
+ (*call_med->med_create_cb)(call_med, call_med->tp_ready,
+ call_med->call->secure_level, NULL);
+
+ if (dlg)
+ pjsip_dlg_dec_lock(dlg);
+ }
+}
+
+/* Deferred callback to notify ICE negotiation failure */
+static void ice_failed_nego_cb(void *user_data)
+{
+ int call_id = (int)(pj_ssize_t)user_data;
+ pjsua_call *call = NULL;
+ pjsip_dialog *dlg = NULL;
+
+ if (acquire_call("ice_failed_nego_cb", call_id,
+ &call, &dlg) != PJ_SUCCESS)
+ {
+ /* Call have been terminated */
+ return;
+ }
+
+ pjsua_var.ua_cfg.cb.on_call_media_state(call_id);
+
+ if (dlg)
+ pjsip_dlg_dec_lock(dlg);
+
+}
+
+/* This callback is called when ICE negotiation completes */
+static void on_ice_complete(pjmedia_transport *tp,
+ pj_ice_strans_op op,
+ pj_status_t result)
+{
+ pjsua_call_media *call_med = (pjsua_call_media*)tp->user_data;
+ pjsua_call *call;
+
+ if (!call_med)
+ return;
+
+ call = call_med->call;
+
+ switch (op) {
+ case PJ_ICE_STRANS_OP_INIT:
+ call_med->tp_result = result;
+ pjsua_schedule_timer2(&ice_init_complete_cb, call_med, 1);
+ break;
+ case PJ_ICE_STRANS_OP_NEGOTIATION:
+ if (result == PJ_SUCCESS) {
+ /* Update RTP address */
+ pjmedia_transport_info tpinfo;
+ pjmedia_transport_info_init(&tpinfo);
+ pjmedia_transport_get_info(call_med->tp, &tpinfo);
+ pj_sockaddr_cp(&call_med->rtp_addr, &tpinfo.sock_info.rtp_addr_name);
+ } else {
+ call_med->state = PJSUA_CALL_MEDIA_ERROR;
+ call_med->dir = PJMEDIA_DIR_NONE;
+ if (call && pjsua_var.ua_cfg.cb.on_call_media_state) {
+ /* Defer the callback to a timer */
+ pjsua_schedule_timer2(&ice_failed_nego_cb,
+ (void*)(pj_ssize_t)call->index, 1);
+ }
+ }
+ /* Check if default ICE transport address is changed */
+ call->reinv_ice_sent = PJ_FALSE;
+ pjsua_call_schedule_reinvite_check(call, 0);
+ break;
+ case PJ_ICE_STRANS_OP_KEEP_ALIVE:
+ if (result != PJ_SUCCESS) {
+ PJ_PERROR(4,(THIS_FILE, result,
+ "ICE keep alive failure for transport %d:%d",
+ call->index, call_med->idx));
+ }
+ if (pjsua_var.ua_cfg.cb.on_call_media_transport_state) {
+ pjsua_med_tp_state_info info;
+
+ pj_bzero(&info, sizeof(info));
+ info.med_idx = call_med->idx;
+ info.state = call_med->tp_st;
+ info.status = result;
+ info.ext_info = &op;
+ (*pjsua_var.ua_cfg.cb.on_call_media_transport_state)(
+ call->index, &info);
+ }
+ if (pjsua_var.ua_cfg.cb.on_ice_transport_error) {
+ pjsua_call_id id = call->index;
+ (*pjsua_var.ua_cfg.cb.on_ice_transport_error)(id, op, result,
+ NULL);
+ }
+ break;
+ }
+}
+
+
+/* Parse "HOST:PORT" format */
+static pj_status_t parse_host_port(const pj_str_t *host_port,
+ pj_str_t *host, pj_uint16_t *port)
+{
+ pj_str_t str_port;
+
+ str_port.ptr = pj_strchr(host_port, ':');
+ if (str_port.ptr != NULL) {
+ int iport;
+
+ host->ptr = host_port->ptr;
+ host->slen = (str_port.ptr - host->ptr);
+ str_port.ptr++;
+ str_port.slen = host_port->slen - host->slen - 1;
+ iport = (int)pj_strtoul(&str_port);
+ if (iport < 1 || iport > 65535)
+ return PJ_EINVAL;
+ *port = (pj_uint16_t)iport;
+ } else {
+ *host = *host_port;
+ *port = 0;
+ }
+
+ return PJ_SUCCESS;
+}
+
+/* Create ICE media transports (when ice is enabled) */
+static pj_status_t create_ice_media_transport(
+ const pjsua_transport_config *cfg,
+ pjsua_call_media *call_med,
+ pj_bool_t async)
+{
+ char stunip[PJ_INET6_ADDRSTRLEN];
+ pjsua_acc_config *acc_cfg;
+ pj_ice_strans_cfg ice_cfg;
+ pjmedia_ice_cb ice_cb;
+ char name[32];
+ unsigned comp_cnt;
+ pj_status_t status;
+
+ acc_cfg = &pjsua_var.acc[call_med->call->acc_id].cfg;
+
+ /* Make sure STUN server resolution has completed */
+ status = resolve_stun_server(PJ_TRUE);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Error resolving STUN server", status);
+ return status;
+ }
+
+ /* Create ICE stream transport configuration */
+ pj_ice_strans_cfg_default(&ice_cfg);
+ pj_stun_config_init(&ice_cfg.stun_cfg, &pjsua_var.cp.factory, 0,
+ pjsip_endpt_get_ioqueue(pjsua_var.endpt),
+ pjsip_endpt_get_timer_heap(pjsua_var.endpt));
+
+ ice_cfg.af = pj_AF_INET();
+ ice_cfg.resolver = pjsua_var.resolver;
+
+ ice_cfg.opt = acc_cfg->ice_cfg.ice_opt;
+
+ /* Configure STUN settings */
+ if (pj_sockaddr_has_addr(&pjsua_var.stun_srv)) {
+ pj_sockaddr_print(&pjsua_var.stun_srv, stunip, sizeof(stunip), 0);
+ ice_cfg.stun.server = pj_str(stunip);
+ ice_cfg.stun.port = pj_sockaddr_get_port(&pjsua_var.stun_srv);
+ }
+ if (acc_cfg->ice_cfg.ice_max_host_cands >= 0)
+ ice_cfg.stun.max_host_cands = acc_cfg->ice_cfg.ice_max_host_cands;
+
+ /* Copy binding port setting to STUN setting */
+ pj_sockaddr_init(ice_cfg.af, &ice_cfg.stun.cfg.bound_addr,
+ &cfg->bound_addr, (pj_uint16_t)cfg->port);
+ ice_cfg.stun.cfg.port_range = (pj_uint16_t)cfg->port_range;
+ if (cfg->port != 0 && ice_cfg.stun.cfg.port_range == 0)
+ ice_cfg.stun.cfg.port_range =
+ (pj_uint16_t)(pjsua_var.ua_cfg.max_calls * 10);
+
+ /* Copy QoS setting to STUN setting */
+ ice_cfg.stun.cfg.qos_type = cfg->qos_type;
+ pj_memcpy(&ice_cfg.stun.cfg.qos_params, &cfg->qos_params,
+ sizeof(cfg->qos_params));
+
+ /* Configure TURN settings */
+ if (acc_cfg->turn_cfg.enable_turn) {
+ status = parse_host_port(&acc_cfg->turn_cfg.turn_server,
+ &ice_cfg.turn.server,
+ &ice_cfg.turn.port);
+ if (status != PJ_SUCCESS || ice_cfg.turn.server.slen == 0) {
+ PJ_LOG(1,(THIS_FILE, "Invalid TURN server setting"));
+ return PJ_EINVAL;
+ }
+ if (ice_cfg.turn.port == 0)
+ ice_cfg.turn.port = 3479;
+ ice_cfg.turn.conn_type = acc_cfg->turn_cfg.turn_conn_type;
+ pj_memcpy(&ice_cfg.turn.auth_cred,
+ &acc_cfg->turn_cfg.turn_auth_cred,
+ sizeof(ice_cfg.turn.auth_cred));
+
+ /* Copy QoS setting to TURN setting */
+ ice_cfg.turn.cfg.qos_type = cfg->qos_type;
+ pj_memcpy(&ice_cfg.turn.cfg.qos_params, &cfg->qos_params,
+ sizeof(cfg->qos_params));
+
+ /* Copy binding port setting to TURN setting */
+ pj_sockaddr_init(ice_cfg.af, &ice_cfg.turn.cfg.bound_addr,
+ &cfg->bound_addr, (pj_uint16_t)cfg->port);
+ ice_cfg.turn.cfg.port_range = (pj_uint16_t)cfg->port_range;
+ if (cfg->port != 0 && ice_cfg.turn.cfg.port_range == 0)
+ ice_cfg.turn.cfg.port_range =
+ (pj_uint16_t)(pjsua_var.ua_cfg.max_calls * 10);
+ }
+
+ /* Configure packet size for STUN and TURN sockets */
+ ice_cfg.stun.cfg.max_pkt_size = PJMEDIA_MAX_MRU;
+ ice_cfg.turn.cfg.max_pkt_size = PJMEDIA_MAX_MRU;
+
+ pj_bzero(&ice_cb, sizeof(pjmedia_ice_cb));
+ ice_cb.on_ice_complete = &on_ice_complete;
+ pj_ansi_snprintf(name, sizeof(name), "icetp%02d", call_med->idx);
+ call_med->tp_ready = PJ_EPENDING;
+
+ comp_cnt = 1;
+ if (PJMEDIA_ADVERTISE_RTCP && !acc_cfg->ice_cfg.ice_no_rtcp)
+ ++comp_cnt;
+
+ status = pjmedia_ice_create3(pjsua_var.med_endpt, name, comp_cnt,
+ &ice_cfg, &ice_cb, 0, call_med,
+ &call_med->tp);
+ if (status != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Unable to create ICE media transport",
+ status);
+ goto on_error;
+ }
+
+ /* Wait until transport is initialized, or time out */
+ if (!async) {
+ pj_bool_t has_pjsua_lock = PJSUA_LOCK_IS_LOCKED();
+ if (has_pjsua_lock)
+ PJSUA_UNLOCK();
+ while (call_med->tp_ready == PJ_EPENDING) {
+ pjsua_handle_events(100);
+ }
+ if (has_pjsua_lock)
+ PJSUA_LOCK();
+ }
+
+ if (async && call_med->tp_ready == PJ_EPENDING) {
+ return PJ_EPENDING;
+ } else if (call_med->tp_ready != PJ_SUCCESS) {
+ pjsua_perror(THIS_FILE, "Error initializing ICE media transport",
+ call_med->tp_ready);
+ status = call_med->tp_ready;
+ goto on_error;
+ }
+
+ pjmedia_transport_simulate_lost(call_med->tp, PJMEDIA_DIR_ENCODING,
+ pjsua_var.media_cfg.tx_drop_pct);
+
+ pjmedia_transport_simulate_lost(call_med->tp, PJMEDIA_DIR_DECODING,
+ pjsua_var.media_cfg.rx_drop_pct);
+
+ return PJ_SUCCESS;
+
+on_error:
+ if (call_med->tp != NULL) {
+ pjmedia_transport_close(call_med->tp);
+ call_med->tp = NULL;
+ }
+
+ return status;
+}
+
+#if DISABLED_FOR_TICKET_1185
+/* Create ICE media transports (when ice is enabled) */
+static pj_status_t create_ice_media_transports(pjsua_transport_config *cfg)
+{
+ unsigned i;
+ pj_status_t status;
+
+ for (i=0; i < pjsua_var.ua_cfg.max_calls; ++i) {
+ pjsua_call *call = &pjsua_var.calls[i];
+ unsigned strm_idx;
+
+ for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) {
+ pjsua_call_media *call_med = &call->media[strm_idx];
+
+ status = create_ice_media_transport(cfg, call_med);
+ if (status != PJ_SUCCESS)
+ goto on_error;
+ }
+ }
+
+ return PJ_SUCCESS;
+
+on_error:
+ for (i=0; i < pjsua_var.ua_cfg.max_calls; ++i) {
+ pjsua_call *call = &pjsua_var.calls[i];
+ unsigned strm_idx;
+
+ for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) {
+ pjsua_call_media *call_med = &call->media[strm_idx];
+
+ if (call_med->tp) {
+ pjmedia_transport_close(call_med->tp);
+ call_med->tp = NULL;
+ }
+ }
+ }
+ return status;
+}
+#endif
+
+#if DISABLED_FOR_TICKET_1185
+/*
+ * Create media transports for all the calls. This function creates
+ * one UDP media transport for each call.
+ */
+PJ_DEF(pj_status_t) pjsua_media_transports_create(
+ const pjsua_transport_config *app_cfg)
+{
+ pjsua_transport_config cfg;
+ unsigned i;
+ pj_status_t status;
+
+
+ /* Make sure pjsua_init() has been called */
+ PJ_ASSERT_RETURN(pjsua_var.ua_cfg.max_calls>0, PJ_EINVALIDOP);
+
+ PJSUA_LOCK();
+
+ /* Delete existing media transports */
+ for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) {
+ pjsua_call *call = &pjsua_var.calls[i];
+ unsigned strm_idx;
+
+ for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) {
+ pjsua_call_media *call_med = &call->media[strm_idx];
+
+ if (call_med->tp && call_med->tp_auto_del) {
+ pjmedia_transport_close(call_med->tp);
+ call_med->tp = NULL;
+ call_med->tp_orig = NULL;
+ }
+ }
+ }
+
+ /* Copy config */
+ pjsua_transport_config_dup(pjsua_var.pool, &cfg, app_cfg);
+
+ /* Create the transports */
+ if (pjsua_var.ice_cfg.enable_ice) {
+ status = create_ice_media_transports(&cfg);
+ } else {
+ status = create_udp_media_transports(&cfg);
+ }
+
+ /* Set media transport auto_delete to True */
+ for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) {
+ pjsua_call *call = &pjsua_var.calls[i];
+ unsigned strm_idx;
+
+ for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) {
+ pjsua_call_media *call_med = &call->media[strm_idx];
+
+ call_med->tp_auto_del = PJ_TRUE;
+ }
+ }
+
+ PJSUA_UNLOCK();
+
+ return status;
+}
+
+/*
+ * Attach application's created media transports.
+ */
+PJ_DEF(pj_status_t) pjsua_media_transports_attach(pjsua_media_transport tp[],
+ unsigned count,
+ pj_bool_t auto_delete)
+{
+ unsigned i;
+
+ PJ_ASSERT_RETURN(tp && count==pjsua_var.ua_cfg.max_calls, PJ_EINVAL);
+
+ /* Assign the media transports */
+ for (i=0; i<pjsua_var.ua_cfg.max_calls; ++i) {
+ pjsua_call *call = &pjsua_var.calls[i];
+ unsigned strm_idx;
+
+ for (strm_idx=0; strm_idx < call->med_cnt; ++strm_idx) {
+ pjsua_call_media *call_med = &call->media[strm_idx];
+
+ if (call_med->tp && call_med->tp_auto_del) {
+ pjmedia_transport_close(call_med->tp);
+ call_med->tp = NULL;
+ call_med->tp_orig = NULL;
+ }
+ }
+
+ PJ_TODO(remove_pjsua_media_transports_attach);
+
+ call->media[0].tp = tp[i].transport;
+ call->media[0].tp_auto_del = auto_delete;
+ }
+
+ return PJ_SUCCESS;
+}
+#endif
+
+/* Go through the list of media in the SDP, find acceptable media, and
+ * sort them based on the "quality" of the media, and store the indexes
+ * in the specified array. Media with the best quality will be listed
+ * first in the array. The quality factors considered currently is
+ * encryption.
+ */
+static void sort_media(const pjmedia_sdp_session *sdp,
+ const pj_str_t *type,
+ pjmedia_srtp_use use_srtp,
+ pj_uint8_t midx[],
+ unsigned *p_count,
+ unsigned *p_total_count)
+{
+ unsigned i;
+ unsigned count = 0;
+ int score[PJSUA_MAX_CALL_MEDIA];
+
+ pj_assert(*p_count >= PJSUA_MAX_CALL_MEDIA);
+ pj_assert(*p_total_count >= PJSUA_MAX_CALL_MEDIA);
+
+ *p_count = 0;
+ *p_total_count = 0;
+ for (i=0; i<PJSUA_MAX_CALL_MEDIA; ++i)
+ score[i] = 1;
+
+ /* Score each media */
+ for (i=0; i<sdp->media_count && count<PJSUA_MAX_CALL_MEDIA; ++i) {
+ const pjmedia_sdp_media *m = sdp->media[i];
+ const pjmedia_sdp_conn *c;
+
+ /* Skip different media */
+ if (pj_stricmp(&m->desc.media, type) != 0) {
+ score[count++] = -22000;
+ continue;
+ }
+
+ c = m->conn? m->conn : sdp->conn;
+
+ /* Supported transports */
+ if (pj_stricmp2(&m->desc.transport, "RTP/SAVP")==0) {
+ switch (use_srtp) {
+ case PJMEDIA_SRTP_MANDATORY:
+ case PJMEDIA_SRTP_OPTIONAL:
+ ++score[i];
+ break;
+ case PJMEDIA_SRTP_DISABLED:
+ //--score[i];
+ score[i] -= 5;
+ break;
+ }
+ } else if (pj_stricmp2(&m->desc.transport, "RTP/AVP")==0) {
+ switch (use_srtp) {
+ case PJMEDIA_SRTP_MANDATORY:
+ //--score[i];
+ score[i] -= 5;
+ break;
+ case PJMEDIA_SRTP_OPTIONAL:
+ /* No change in score */
+ break;
+ case PJMEDIA_SRTP_DISABLED:
+ ++score[i];
+ break;
+ }
+ } else {
+ score[i] -= 10;
+ }
+
+ /* Is media disabled? */
+ if (m->desc.port == 0)
+ score[i] -= 10;
+
+ /* Is media inactive? */
+ if (pjmedia_sdp_media_find_attr2(m, "inactive", NULL) ||
+ pj_strcmp2(&c->addr, "0.0.0.0") == 0)
+ {
+ //score[i] -= 10;
+ score[i] -= 1;
+ }
+
+ ++count;
+ }
+
+ /* Created sorted list based on quality */
+ for (i=0; i<count; ++i) {
+ unsigned j;
+ int best = 0;
+
+ for (j=1; j<count; ++j) {
+ if (score[j] > score[best])
+ best = j;
+ }
+ /* Don't put media with negative score, that media is unacceptable
+ * for us.
+ */
+ midx[i] = (pj_uint8_t)best;
+ if (score[best] >= 0)
+ (*p_count)++;
+ if (score[best] > -22000)
+ (*p_total_count)++;
+
+ score[best] = -22000;
+
+ }
+}
+
+/* Callback to receive media events */
+pj_status_t call_media_on_event(pjmedia_event *event,
+ void *user_data)
+{
+ pjsua_call_media *call_med = (pjsua_call_media*)user_data;
+ pjsua_call *call = call_med->call;
+ pj_status_t status = PJ_SUCCESS;
+
+ switch(event->type) {
+ case PJMEDIA_EVENT_KEYFRAME_MISSING:
+ if (call->opt.req_keyframe_method & PJSUA_VID_REQ_KEYFRAME_SIP_INFO)
+ {
+ pj_timestamp now;
+
+ pj_get_timestamp(&now);
+ if (pj_elapsed_msec(&call_med->last_req_keyframe, &now) >=
+ PJSUA_VID_REQ_KEYFRAME_INTERVAL)
+ {
+ pjsua_msg_data msg_data;
+ const pj_str_t SIP_INFO = {"INFO", 4};
+ const char *BODY_TYPE = "application/media_control+xml";
+ const char *BODY =
+ "<?xml version=\"1.0\" encoding=\"utf-8\" ?>"
+ "<media_control><vc_primitive><to_encoder>"
+ "<picture_fast_update/>"
+ "</to_encoder></vc_primitive></media_control>";
+
+ PJ_LOG(4,(THIS_FILE,
+ "Sending video keyframe request via SIP INFO"));
+
+ pjsua_msg_data_init(&msg_data);
+ pj_cstr(&msg_data.content_type, BODY_TYPE);
+ pj_cstr(&msg_data.msg_body, BODY);
+ status = pjsua_call_send_request(call->index, &SIP_INFO,
+ &msg_data);
+ if (status != PJ_SUCCESS) {
+ pj_perror(3, THIS_FILE, status,
+ "Failed requesting keyframe via SIP INFO");
+ } else {
+ call_med->last_req_keyframe = now;
+ }
+ }
+ }
+ break;
+
+ default:
+ break;
+ }
+
+ if (pjsua_var.ua_cfg.cb.on_call_media_event && call) {
+ (*pjsua_var.ua_cfg.cb.on_call_media_event)(call->index,
+ call_med->idx, event);
+ }
+
+ return status;
+}
+
+/* Set media transport state and notify the application via the callback. */
+void pjsua_set_media_tp_state(pjsua_call_media *call_med,
+ pjsua_med_tp_st tp_st)
+{
+ if (pjsua_var.ua_cfg.cb.on_call_media_transport_state &&
+ call_med->tp_st != tp_st)
+ {
+ pjsua_med_tp_state_info info;
+
+ pj_bzero(&info, sizeof(info));
+ info.med_idx = call_med->idx;
+ info.state = tp_st;
+ info.status = call_med->tp_ready;
+ (*pjsua_var.ua_cfg.cb.on_call_media_transport_state)(
+ call_med->call->index, &info);
+ }
+
+ call_med->tp_st = tp_st;
+}
+
+/* Callback to resume pjsua_call_media_init() after media transport
+ * creation is completed.
+ */
+static pj_status_t call_media_init_cb(pjsua_call_media *call_med,
+ pj_status_t status,
+ int security_level,
+ int *sip_err_code)
+{
+ pjsua_acc *acc = &pjsua_var.acc[call_med->call->acc_id];
+ pjmedia_transport_info tpinfo;
+ int err_code = 0;
+
+ if (status != PJ_SUCCESS)
+ goto on_return;
+
+ pjmedia_transport_simulate_lost(call_med->tp, PJMEDIA_DIR_ENCODING,
+ pjsua_var.media_cfg.tx_drop_pct);
+
+ pjmedia_transport_simulate_lost(call_med->tp, PJMEDIA_DIR_DECODING,
+ pjsua_var.media_cfg.rx_drop_pct);
+
+ if (call_med->tp_st == PJSUA_MED_TP_CREATING)
+ pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_IDLE);
+
+ if (!call_med->tp_orig &&
+ pjsua_var.ua_cfg.cb.on_create_media_transport)
+ {
+ call_med->use_custom_med_tp = PJ_TRUE;
+ } else
+ call_med->use_custom_med_tp = PJ_FALSE;
+
+#if defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
+ /* This function may be called when SRTP transport already exists
+ * (e.g: in re-invite, update), don't need to destroy/re-create.
+ */
+ if (!call_med->tp_orig) {
+ pjmedia_srtp_setting srtp_opt;
+ pjmedia_transport *srtp = NULL;
+
+ /* Check if SRTP requires secure signaling */
+ if (acc->cfg.use_srtp != PJMEDIA_SRTP_DISABLED) {
+ if (security_level < acc->cfg.srtp_secure_signaling) {
+ err_code = PJSIP_SC_NOT_ACCEPTABLE;
+ status = PJSIP_ESESSIONINSECURE;
+ goto on_return;
+ }
+ }
+
+ /* Always create SRTP adapter */
+ pjmedia_srtp_setting_default(&srtp_opt);
+ srtp_opt.close_member_tp = PJ_TRUE;
+
+ /* If media session has been ever established, let's use remote's
+ * preference in SRTP usage policy, especially when it is stricter.
+ */
+ if (call_med->rem_srtp_use > acc->cfg.use_srtp)
+ srtp_opt.use = call_med->rem_srtp_use;
+ else
+ srtp_opt.use = acc->cfg.use_srtp;
+
+ status = pjmedia_transport_srtp_create(pjsua_var.med_endpt,
+ call_med->tp,
+ &srtp_opt, &srtp);
+ if (status != PJ_SUCCESS) {
+ err_code = PJSIP_SC_INTERNAL_SERVER_ERROR;
+ goto on_return;
+ }
+
+ /* Set SRTP as current media transport */
+ call_med->tp_orig = call_med->tp;
+ call_med->tp = srtp;
+ }
+#else
+ call_med->tp_orig = call_med->tp;
+ PJ_UNUSED_ARG(security_level);
+#endif
+
+
+ pjmedia_transport_info_init(&tpinfo);
+ pjmedia_transport_get_info(call_med->tp, &tpinfo);
+
+ pj_sockaddr_cp(&call_med->rtp_addr, &tpinfo.sock_info.rtp_addr_name);
+
+
+on_return:
+ if (status != PJ_SUCCESS && call_med->tp) {
+ pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_NULL);
+ pjmedia_transport_close(call_med->tp);
+ call_med->tp = NULL;
+ }
+
+ if (sip_err_code)
+ *sip_err_code = err_code;
+
+ if (call_med->med_init_cb) {
+ pjsua_med_tp_state_info info;
+
+ pj_bzero(&info, sizeof(info));
+ info.status = status;
+ info.state = call_med->tp_st;
+ info.med_idx = call_med->idx;
+ info.sip_err_code = err_code;
+ (*call_med->med_init_cb)(call_med->call->index, &info);
+ }
+
+ return status;
+}
+
+/* Initialize the media line */
+pj_status_t pjsua_call_media_init(pjsua_call_media *call_med,
+ pjmedia_type type,
+ const pjsua_transport_config *tcfg,
+ int security_level,
+ int *sip_err_code,
+ pj_bool_t async,
+ pjsua_med_tp_state_cb cb)
+{
+ pj_status_t status = PJ_SUCCESS;
+
+ /*
+ * Note: this function may be called when the media already exists
+ * (e.g. in reinvites, updates, etc.)
+ */
+ call_med->type = type;
+
+ /* Create the media transport for initial call. Here are the possible
+ * media transport state and the action needed:
+ * - PJSUA_MED_TP_NULL or call_med->tp==NULL, create one.
+ * - PJSUA_MED_TP_RUNNING, do nothing.
+ * - PJSUA_MED_TP_DISABLED, re-init (media_create(), etc). Currently,
+ * this won't happen as media_channel_update() will always clean up
+ * the unused transport of a disabled media.
+ */
+ if (call_med->tp == NULL) {
+#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
+ /* While in initial call, set default video devices */
+ if (type == PJMEDIA_TYPE_VIDEO) {
+ status = pjsua_vid_channel_init(call_med);
+ if (status != PJ_SUCCESS)
+ return status;
+ }
+#endif
+
+ pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_CREATING);
+
+ if (pjsua_var.acc[call_med->call->acc_id].cfg.ice_cfg.enable_ice) {
+ status = create_ice_media_transport(tcfg, call_med, async);
+ if (async && status == PJ_EPENDING) {
+ /* We will resume call media initialization in the
+ * on_ice_complete() callback.
+ */
+ call_med->med_create_cb = &call_media_init_cb;
+ call_med->med_init_cb = cb;
+
+ return PJ_EPENDING;
+ }
+ } else {
+ status = create_udp_media_transport(tcfg, call_med);
+ }
+
+ if (status != PJ_SUCCESS) {
+ PJ_PERROR(1,(THIS_FILE, status, "Error creating media transport"));
+ return status;
+ }
+
+ /* Media transport creation completed immediately, so
+ * we don't need to call the callback.
+ */
+ call_med->med_init_cb = NULL;
+
+ } else if (call_med->tp_st == PJSUA_MED_TP_DISABLED) {
+ /* Media is being reenabled. */
+ //pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_IDLE);
+
+ pj_assert(!"Currently no media transport reuse");
+ }
+
+ return call_media_init_cb(call_med, status, security_level,
+ sip_err_code);
+}
+
+/* Callback to resume pjsua_media_channel_init() after media transport
+ * initialization is completed.
+ */
+static pj_status_t media_channel_init_cb(pjsua_call_id call_id,
+ const pjsua_med_tp_state_info *info)
+{
+ pjsua_call *call = &pjsua_var.calls[call_id];
+ pj_status_t status = (info? info->status : PJ_SUCCESS);
+ unsigned mi;
+
+ if (info) {
+ pj_mutex_lock(call->med_ch_mutex);
+
+ /* Set the callback to NULL to indicate that the async operation
+ * has completed.
+ */
+ call->media_prov[info->med_idx].med_init_cb = NULL;
+
+ /* In case of failure, save the information to be returned
+ * by the last media transport to finish.
+ */
+ if (info->status != PJ_SUCCESS)
+ pj_memcpy(&call->med_ch_info, info, sizeof(*info));
+
+ /* Check whether all the call's medias have finished calling their
+ * callbacks.
+ */
+ for (mi=0; mi < call->med_prov_cnt; ++mi) {
+ pjsua_call_media *call_med = &call->media_prov[mi];
+
+ if (call_med->med_init_cb) {
+ pj_mutex_unlock(call->med_ch_mutex);
+ return PJ_SUCCESS;
+ }
+
+ if (call_med->tp_ready != PJ_SUCCESS)
+ status = call_med->tp_ready;
+ }
+
+ /* OK, we are called by the last media transport finished. */
+ pj_mutex_unlock(call->med_ch_mutex);
+ }
+
+ if (call->med_ch_mutex) {
+ pj_mutex_destroy(call->med_ch_mutex);
+ call->med_ch_mutex = NULL;
+ }
+
+ if (status != PJ_SUCCESS) {
+ if (call->med_ch_info.status == PJ_SUCCESS) {
+ call->med_ch_info.status = status;
+ call->med_ch_info.sip_err_code = PJSIP_SC_TEMPORARILY_UNAVAILABLE;
+ }
+ pjsua_media_prov_clean_up(call_id);
+ goto on_return;
+ }
+
+ /* Tell the media transport of a new offer/answer session */
+ for (mi=0; mi < call->med_prov_cnt; ++mi) {
+ pjsua_call_media *call_med = &call->media_prov[mi];
+
+ /* Note: tp may be NULL if this media line is disabled */
+ if (call_med->tp && call_med->tp_st == PJSUA_MED_TP_IDLE) {
+ pj_pool_t *tmp_pool = call->async_call.pool_prov;
+
+ if (!tmp_pool) {
+ tmp_pool = (call->inv? call->inv->pool_prov:
+ call->async_call.dlg->pool);
+ }
+
+ if (call_med->use_custom_med_tp) {
+ unsigned custom_med_tp_flags = PJSUA_MED_TP_CLOSE_MEMBER;
+
+ /* Use custom media transport returned by the application */
+ call_med->tp =
+ (*pjsua_var.ua_cfg.cb.on_create_media_transport)
+ (call_id, mi, call_med->tp,
+ custom_med_tp_flags);
+ if (!call_med->tp) {
+ status =
+ PJSIP_ERRNO_FROM_SIP_STATUS(PJSIP_SC_TEMPORARILY_UNAVAILABLE);
+ }
+ }
+
+ if (call_med->tp) {
+ status = pjmedia_transport_media_create(
+ call_med->tp, tmp_pool,
+ 0, call->async_call.rem_sdp, mi);
+ }
+ if (status != PJ_SUCCESS) {
+ call->med_ch_info.status = status;
+ call->med_ch_info.med_idx = mi;
+ call->med_ch_info.state = call_med->tp_st;
+ call->med_ch_info.sip_err_code = PJSIP_SC_TEMPORARILY_UNAVAILABLE;
+ pjsua_media_prov_clean_up(call_id);
+ goto on_return;
+ }
+
+ pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_INIT);
+ }
+ }
+
+ call->med_ch_info.status = PJ_SUCCESS;
+
+on_return:
+ if (call->med_ch_cb)
+ (*call->med_ch_cb)(call->index, &call->med_ch_info);
+
+ return status;
+}
+
+
+/* Clean up media transports in provisional media that is not used
+ * by call media.
+ */
+static void media_prov_clean_up(pjsua_call_id call_id, int idx)
+{
+ pjsua_call *call = &pjsua_var.calls[call_id];
+ unsigned i;
+
+ for (i = idx; i < call->med_prov_cnt; ++i) {
+ pjsua_call_media *call_med = &call->media_prov[i];
+ unsigned j;
+ pj_bool_t used = PJ_FALSE;
+
+ if (call_med->tp == NULL)
+ continue;
+
+ for (j = 0; j < call->med_cnt; ++j) {
+ if (call->media[j].tp == call_med->tp) {
+ used = PJ_TRUE;
+ break;
+ }
+ }
+
+ if (!used) {
+ if (call_med->tp_st > PJSUA_MED_TP_IDLE) {
+ pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_IDLE);
+ pjmedia_transport_media_stop(call_med->tp);
+ }
+ pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_NULL);
+ pjmedia_transport_close(call_med->tp);
+ call_med->tp = call_med->tp_orig = NULL;
+ }
+ }
+}
+
+void pjsua_media_prov_clean_up(pjsua_call_id call_id)
+{
+ media_prov_clean_up(call_id, 0);
+}
+
+
+pj_status_t pjsua_media_channel_init(pjsua_call_id call_id,
+ pjsip_role_e role,
+ int security_level,
+ pj_pool_t *tmp_pool,
+ const pjmedia_sdp_session *rem_sdp,
+ int *sip_err_code,
+ pj_bool_t async,
+ pjsua_med_tp_state_cb cb)
+{
+ const pj_str_t STR_AUDIO = { "audio", 5 };
+ const pj_str_t STR_VIDEO = { "video", 5 };
+ pjsua_call *call = &pjsua_var.calls[call_id];
+ pjsua_acc *acc = &pjsua_var.acc[call->acc_id];
+ pj_uint8_t maudidx[PJSUA_MAX_CALL_MEDIA];
+ unsigned maudcnt = PJ_ARRAY_SIZE(maudidx);
+ unsigned mtotaudcnt = PJ_ARRAY_SIZE(maudidx);
+ pj_uint8_t mvididx[PJSUA_MAX_CALL_MEDIA];
+ unsigned mvidcnt = PJ_ARRAY_SIZE(mvididx);
+ unsigned mtotvidcnt = PJ_ARRAY_SIZE(mvididx);
+ unsigned mi;
+ pj_bool_t pending_med_tp = PJ_FALSE;
+ pj_bool_t reinit = PJ_FALSE;
+ pj_status_t status;
+
+ PJ_UNUSED_ARG(role);
+
+ /*
+ * Note: this function may be called when the media already exists
+ * (e.g. in reinvites, updates, etc).
+ */
+
+ if (pjsua_get_state() != PJSUA_STATE_RUNNING) {
+ if (sip_err_code) *sip_err_code = PJSIP_SC_SERVICE_UNAVAILABLE;
+ return PJ_EBUSY;
+ }
+
+ if (async) {
+ pj_pool_t *tmppool = (call->inv? call->inv->pool_prov:
+ call->async_call.dlg->pool);
+
+ status = pj_mutex_create_simple(tmppool, NULL, &call->med_ch_mutex);
+ if (status != PJ_SUCCESS)
+ return status;
+ }
+
+ if (call->inv && call->inv->state == PJSIP_INV_STATE_CONFIRMED)
+ reinit = PJ_TRUE;
+
+ PJ_LOG(4,(THIS_FILE, "Call %d: %sinitializing media..",
+ call_id, (reinit?"re-":"") ));
+
+ pj_log_push_indent();
+
+ /* Init provisional media state */
+ if (call->med_cnt == 0) {
+ /* New media session, just copy whole from call media state. */
+ pj_memcpy(call->media_prov, call->media, sizeof(call->media));
+ } else {
+ /* Clean up any unused transports. Note that when local SDP reoffer
+ * is rejected by remote, there may be any initialized transports that
+ * are not used by call media and currently there is no notification
+ * from PJSIP level regarding the reoffer rejection.
+ */
+ pjsua_media_prov_clean_up(call_id);
+
+ /* Updating media session, copy from call media state. */
+ pj_memcpy(call->media_prov, call->media,
+ sizeof(call->media[0]) * call->med_cnt);
+ }
+ call->med_prov_cnt = call->med_cnt;
+
+#if DISABLED_FOR_TICKET_1185
+ /* Return error if media transport has not been created yet
+ * (e.g. application is starting)
+ */
+ for (i=0; i<call->med_cnt; ++i) {
+ if (call->media[i].tp == NULL) {
+ status = PJ_EBUSY;
+ goto on_error;
+ }
+ }
+#endif
+
+ /* Get media count for each media type */
+ if (rem_sdp) {
+ sort_media(rem_sdp, &STR_AUDIO, acc->cfg.use_srtp,
+ maudidx, &maudcnt, &mtotaudcnt);
+ if (maudcnt==0) {
+ /* Expecting audio in the offer */
+ if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE;
+ status = PJSIP_ERRNO_FROM_SIP_STATUS(PJSIP_SC_NOT_ACCEPTABLE_HERE);
+ goto on_error;
+ }
+
+#if PJMEDIA_HAS_VIDEO
+ sort_media(rem_sdp, &STR_VIDEO, acc->cfg.use_srtp,
+ mvididx, &mvidcnt, &mtotvidcnt);
+#else
+ mvidcnt = mtotvidcnt = 0;
+ PJ_UNUSED_ARG(STR_VIDEO);
+#endif
+
+ /* Update media count only when remote add any media, this media count
+ * must never decrease. Also note that we shouldn't apply the media
+ * count setting (of the call setting) before the SDP negotiation.
+ */
+ if (call->med_prov_cnt < rem_sdp->media_count)
+ call->med_prov_cnt = PJ_MIN(rem_sdp->media_count,
+ PJSUA_MAX_CALL_MEDIA);
+
+ call->rem_offerer = PJ_TRUE;
+ call->rem_aud_cnt = maudcnt;
+ call->rem_vid_cnt = mvidcnt;
+
+ } else {
+
+ /* If call already established, calculate media count from current
+ * local active SDP and call setting. Otherwise, calculate media
+ * count from the call setting only.
+ */
+ if (reinit) {
+ const pjmedia_sdp_session *sdp;
+
+ status = pjmedia_sdp_neg_get_active_local(call->inv->neg, &sdp);
+ pj_assert(status == PJ_SUCCESS);
+
+ sort_media(sdp, &STR_AUDIO, acc->cfg.use_srtp,
+ maudidx, &maudcnt, &mtotaudcnt);
+ pj_assert(maudcnt > 0);
+
+ sort_media(sdp, &STR_VIDEO, acc->cfg.use_srtp,
+ mvididx, &mvidcnt, &mtotvidcnt);
+
+ /* Call setting may add or remove media. Adding media is done by
+ * enabling any disabled/port-zeroed media first, then adding new
+ * media whenever needed. Removing media is done by disabling
+ * media with the lowest 'quality'.
+ */
+
+ /* Check if we need to add new audio */
+ if (maudcnt < call->opt.aud_cnt &&
+ mtotaudcnt < call->opt.aud_cnt)
+ {
+ for (mi = 0; mi < call->opt.aud_cnt - mtotaudcnt; ++mi)
+ maudidx[maudcnt++] = (pj_uint8_t)call->med_prov_cnt++;
+
+ mtotaudcnt = call->opt.aud_cnt;
+ }
+ maudcnt = call->opt.aud_cnt;
+
+ /* Check if we need to add new video */
+ if (mvidcnt < call->opt.vid_cnt &&
+ mtotvidcnt < call->opt.vid_cnt)
+ {
+ for (mi = 0; mi < call->opt.vid_cnt - mtotvidcnt; ++mi)
+ mvididx[mvidcnt++] = (pj_uint8_t)call->med_prov_cnt++;
+
+ mtotvidcnt = call->opt.vid_cnt;
+ }
+ mvidcnt = call->opt.vid_cnt;
+
+ } else {
+
+ maudcnt = mtotaudcnt = call->opt.aud_cnt;
+ for (mi=0; mi<maudcnt; ++mi) {
+ maudidx[mi] = (pj_uint8_t)mi;
+ }
+ mvidcnt = mtotvidcnt = call->opt.vid_cnt;
+ for (mi=0; mi<mvidcnt; ++mi) {
+ mvididx[mi] = (pj_uint8_t)(maudcnt + mi);
+ }
+ call->med_prov_cnt = maudcnt + mvidcnt;
+
+ /* Need to publish supported media? */
+ if (call->opt.flag & PJSUA_CALL_INCLUDE_DISABLED_MEDIA) {
+ if (mtotaudcnt == 0) {
+ mtotaudcnt = 1;
+ maudidx[0] = (pj_uint8_t)call->med_prov_cnt++;
+ }
+#if PJMEDIA_HAS_VIDEO
+ if (mtotvidcnt == 0) {
+ mtotvidcnt = 1;
+ mvididx[0] = (pj_uint8_t)call->med_prov_cnt++;
+ }
+#endif
+ }
+ }
+
+ call->rem_offerer = PJ_FALSE;
+ }
+
+ if (call->med_prov_cnt == 0) {
+ /* Expecting at least one media */
+ if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE_HERE;
+ status = PJSIP_ERRNO_FROM_SIP_STATUS(PJSIP_SC_NOT_ACCEPTABLE_HERE);
+ goto on_error;
+ }
+
+ if (async) {
+ call->med_ch_cb = cb;
+ }
+
+ if (rem_sdp) {
+ call->async_call.rem_sdp =
+ pjmedia_sdp_session_clone(call->inv->pool_prov, rem_sdp);
+ } else {
+ call->async_call.rem_sdp = NULL;
+ }
+
+ call->async_call.pool_prov = tmp_pool;
+
+ /* Initialize each media line */
+ for (mi=0; mi < call->med_prov_cnt; ++mi) {
+ pjsua_call_media *call_med = &call->media_prov[mi];
+ pj_bool_t enabled = PJ_FALSE;
+ pjmedia_type media_type = PJMEDIA_TYPE_UNKNOWN;
+
+ if (pj_memchr(maudidx, mi, mtotaudcnt * sizeof(maudidx[0]))) {
+ media_type = PJMEDIA_TYPE_AUDIO;
+ if (call->opt.aud_cnt &&
+ pj_memchr(maudidx, mi, maudcnt * sizeof(maudidx[0])))
+ {
+ enabled = PJ_TRUE;
+ }
+ } else if (pj_memchr(mvididx, mi, mtotvidcnt * sizeof(mvididx[0]))) {
+ media_type = PJMEDIA_TYPE_VIDEO;
+ if (call->opt.vid_cnt &&
+ pj_memchr(mvididx, mi, mvidcnt * sizeof(mvididx[0])))
+ {
+ enabled = PJ_TRUE;
+ }
+ }
+
+ if (enabled) {
+ status = pjsua_call_media_init(call_med, media_type,
+ &acc->cfg.rtp_cfg,
+ security_level, sip_err_code,
+ async,
+ (async? &media_channel_init_cb:
+ NULL));
+ if (status == PJ_EPENDING) {
+ pending_med_tp = PJ_TRUE;
+ } else if (status != PJ_SUCCESS) {
+ if (pending_med_tp) {
+ /* Save failure information. */
+ call_med->tp_ready = status;
+ pj_bzero(&call->med_ch_info, sizeof(call->med_ch_info));
+ call->med_ch_info.status = status;
+ call->med_ch_info.state = call_med->tp_st;
+ call->med_ch_info.med_idx = call_med->idx;
+ if (sip_err_code)
+ call->med_ch_info.sip_err_code = *sip_err_code;
+
+ /* We will return failure in the callback later. */
+ return PJ_EPENDING;
+ }
+
+ pjsua_media_prov_clean_up(call_id);
+ goto on_error;
+ }
+ } else {
+ /* By convention, the media is disabled if transport is NULL
+ * or transport state is PJSUA_MED_TP_DISABLED.
+ */
+ if (call_med->tp) {
+ // Don't close transport here, as SDP negotiation has not been
+ // done and stream may be still active. Once SDP negotiation
+ // is done (channel_update() invoked), this transport will be
+ // closed there.
+ //pjmedia_transport_close(call_med->tp);
+ //call_med->tp = NULL;
+ pj_assert(call_med->tp_st == PJSUA_MED_TP_INIT ||
+ call_med->tp_st == PJSUA_MED_TP_RUNNING);
+ pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_DISABLED);
+ }
+
+ /* Put media type just for info */
+ call_med->type = media_type;
+ }
+ }
+
+ call->audio_idx = maudidx[0];
+
+ PJ_LOG(4,(THIS_FILE, "Media index %d selected for audio call %d",
+ call->audio_idx, call->index));
+
+ if (pending_med_tp) {
+ /* We shouldn't use temporary pool anymore. */
+ call->async_call.pool_prov = NULL;
+ /* We have a pending media transport initialization. */
+ pj_log_pop_indent();
+ return PJ_EPENDING;
+ }
+
+ /* Media transport initialization completed immediately, so
+ * we don't need to call the callback.
+ */
+ call->med_ch_cb = NULL;
+
+ status = media_channel_init_cb(call_id, NULL);
+ if (status != PJ_SUCCESS && sip_err_code)
+ *sip_err_code = call->med_ch_info.sip_err_code;
+
+ pj_log_pop_indent();
+ return status;
+
+on_error:
+ if (call->med_ch_mutex) {
+ pj_mutex_destroy(call->med_ch_mutex);
+ call->med_ch_mutex = NULL;
+ }
+
+ pj_log_pop_indent();
+ return status;
+}
+
+
+/* Create SDP based on the current media channel. Note that, this function
+ * will not modify the media channel, so when receiving new offer or
+ * updating media count (via call setting), media channel must be reinit'd
+ * (using pjsua_media_channel_init()) first before calling this function.
+ */
+pj_status_t pjsua_media_channel_create_sdp(pjsua_call_id call_id,
+ pj_pool_t *pool,
+ const pjmedia_sdp_session *rem_sdp,
+ pjmedia_sdp_session **p_sdp,
+ int *sip_err_code)
+{
+ enum { MAX_MEDIA = PJSUA_MAX_CALL_MEDIA };
+ pjmedia_sdp_session *sdp;
+ pj_sockaddr origin;
+ pjsua_call *call = &pjsua_var.calls[call_id];
+ pjmedia_sdp_neg_state sdp_neg_state = PJMEDIA_SDP_NEG_STATE_NULL;
+ unsigned mi;
+ unsigned tot_bandw_tias = 0;
+ pj_status_t status;
+
+ if (pjsua_get_state() != PJSUA_STATE_RUNNING)
+ return PJ_EBUSY;
+
+#if 0
+ // This function should not really change the media channel.
+ if (rem_sdp) {
+ /* If this is a re-offer, let's re-initialize media as remote may
+ * add or remove media
+ */
+ if (call->inv && call->inv->state == PJSIP_INV_STATE_CONFIRMED) {
+ status = pjsua_media_channel_init(call_id, PJSIP_ROLE_UAS,
+ call->secure_level, pool,
+ rem_sdp, sip_err_code,
+ PJ_FALSE, NULL);
+ if (status != PJ_SUCCESS)
+ return status;
+ }
+ } else {
+ /* Audio is first in our offer, by convention */
+ // The audio_idx should not be changed here, as this function may be
+ // called in generating re-offer and the current active audio index
+ // can be anywhere.
+ //call->audio_idx = 0;
+ }
+#endif
+
+#if 0
+ // Since r3512, old-style hold should have got transport, created by
+ // pjsua_media_channel_init() in initial offer/answer or remote reoffer.
+ /* Create media if it's not created. This could happen when call is
+ * currently on-hold (with the old style hold)
+ */
+ if (call->media[call->audio_idx].tp == NULL) {
+ pjsip_role_e role;
+ role = (rem_sdp ? PJSIP_ROLE_UAS : PJSIP_ROLE_UAC);
+ status = pjsua_media_channel_init(call_id, role, call->secure_level,
+ pool, rem_sdp, sip_err_code);
+ if (status != PJ_SUCCESS)
+ return status;
+ }
+#endif
+
+ /* Get SDP negotiator state */
+ if (call->inv && call->inv->neg)
+ sdp_neg_state = pjmedia_sdp_neg_get_state(call->inv->neg);
+
+ /* Get one address to use in the origin field */
+ pj_bzero(&origin, sizeof(origin));
+ for (mi=0; mi<call->med_prov_cnt; ++mi) {
+ pjmedia_transport_info tpinfo;
+
+ if (call->media_prov[mi].tp == NULL)
+ continue;
+
+ pjmedia_transport_info_init(&tpinfo);
+ pjmedia_transport_get_info(call->media_prov[mi].tp, &tpinfo);
+ pj_sockaddr_cp(&origin, &tpinfo.sock_info.rtp_addr_name);
+ break;
+ }
+
+ /* Create the base (blank) SDP */
+ status = pjmedia_endpt_create_base_sdp(pjsua_var.med_endpt, pool, NULL,
+ &origin, &sdp);
+ if (status != PJ_SUCCESS)
+ return status;
+
+ /* Process each media line */
+ for (mi=0; mi<call->med_prov_cnt; ++mi) {
+ pjsua_call_media *call_med = &call->media_prov[mi];
+ pjmedia_sdp_media *m = NULL;
+ pjmedia_transport_info tpinfo;
+ unsigned i;
+
+ if (rem_sdp && mi >= rem_sdp->media_count) {
+ /* Remote might have removed some media lines. */
+ media_prov_clean_up(call->index, rem_sdp->media_count);
+ call->med_prov_cnt = rem_sdp->media_count;
+ break;
+ }
+
+ if (call_med->tp == NULL || call_med->tp_st == PJSUA_MED_TP_DISABLED)
+ {
+ /*
+ * This media is disabled. Just create a valid SDP with zero
+ * port.
+ */
+ if (rem_sdp) {
+ /* Just clone the remote media and deactivate it */
+ m = pjmedia_sdp_media_clone_deactivate(pool,
+ rem_sdp->media[mi]);
+ } else {
+ m = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_media);
+ m->desc.transport = pj_str("RTP/AVP");
+ m->desc.fmt_count = 1;
+
+ switch (call_med->type) {
+ case PJMEDIA_TYPE_AUDIO:
+ m->desc.media = pj_str("audio");
+ m->desc.fmt[0] = pj_str("0");
+ break;
+ case PJMEDIA_TYPE_VIDEO:
+ m->desc.media = pj_str("video");
+ m->desc.fmt[0] = pj_str("31");
+ break;
+ default:
+ /* This must be us generating re-offer, and some unknown
+ * media may exist, so just clone from active local SDP
+ * (and it should have been deactivated already).
+ */
+ pj_assert(call->inv && call->inv->neg &&
+ sdp_neg_state == PJMEDIA_SDP_NEG_STATE_DONE);
+ {
+ const pjmedia_sdp_session *s_;
+ pjmedia_sdp_neg_get_active_local(call->inv->neg, &s_);
+
+ pj_assert(mi < s_->media_count);
+ m = pjmedia_sdp_media_clone(pool, s_->media[mi]);
+ m->desc.port = 0;
+ }
+ break;
+ }
+ }
+
+ /* Add connection line, if none */
+ if (m->conn == NULL && sdp->conn == NULL) {
+ pj_bool_t use_ipv6;
+
+ use_ipv6 = (pjsua_var.acc[call->acc_id].cfg.ipv6_media_use !=
+ PJSUA_IPV6_DISABLED);
+
+ m->conn = PJ_POOL_ZALLOC_T(pool, pjmedia_sdp_conn);
+ m->conn->net_type = pj_str("IN");
+ if (use_ipv6) {
+ m->conn->addr_type = pj_str("IP6");
+ m->conn->addr = pj_str("::1");
+ } else {
+ m->conn->addr_type = pj_str("IP4");
+ m->conn->addr = pj_str("127.0.0.1");
+ }
+ }
+
+ sdp->media[sdp->media_count++] = m;
+ continue;
+ }
+
+ /* Get transport address info */
+ pjmedia_transport_info_init(&tpinfo);
+ pjmedia_transport_get_info(call_med->tp, &tpinfo);
+
+ /* Ask pjmedia endpoint to create SDP media line */
+ switch (call_med->type) {
+ case PJMEDIA_TYPE_AUDIO:
+ status = pjmedia_endpt_create_audio_sdp(pjsua_var.med_endpt, pool,
+ &tpinfo.sock_info, 0, &m);
+ break;
+#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
+ case PJMEDIA_TYPE_VIDEO:
+ status = pjmedia_endpt_create_video_sdp(pjsua_var.med_endpt, pool,
+ &tpinfo.sock_info, 0, &m);
+ break;
+#endif
+ default:
+ pj_assert(!"Invalid call_med media type");
+ return PJ_EBUG;
+ }
+
+ if (status != PJ_SUCCESS)
+ return status;
+
+ sdp->media[sdp->media_count++] = m;
+
+ /* Give to transport */
+ status = pjmedia_transport_encode_sdp(call_med->tp, pool,
+ sdp, rem_sdp, mi);
+ if (status != PJ_SUCCESS) {
+ if (sip_err_code) *sip_err_code = PJSIP_SC_NOT_ACCEPTABLE;
+ return status;
+ }
+
+#if PJSUA_SDP_SESS_HAS_CONN
+ /* Copy c= line of the first media to session level,
+ * if there's none.
+ */
+ if (sdp->conn == NULL) {
+ sdp->conn = pjmedia_sdp_conn_clone(pool, m->conn);
+ }
+#endif
+
+
+ /* Find media bandwidth info */
+ for (i = 0; i < m->bandw_count; ++i) {
+ const pj_str_t STR_BANDW_MODIFIER_TIAS = { "TIAS", 4 };
+ if (!pj_stricmp(&m->bandw[i]->modifier, &STR_BANDW_MODIFIER_TIAS))
+ {
+ tot_bandw_tias += m->bandw[i]->value;
+ break;
+ }
+ }
+ }
+
+ /* Add NAT info in the SDP */
+ if (pjsua_var.ua_cfg.nat_type_in_sdp) {
+ pjmedia_sdp_attr *a;
+ pj_str_t value;
+ char nat_info[80];
+
+ value.ptr = nat_info;
+ if (pjsua_var.ua_cfg.nat_type_in_sdp == 1) {
+ value.slen = pj_ansi_snprintf(nat_info, sizeof(nat_info),
+ "%d", pjsua_var.nat_type);
+ } else {
+ const char *type_name = pj_stun_get_nat_name(pjsua_var.nat_type);
+ value.slen = pj_ansi_snprintf(nat_info, sizeof(nat_info),
+ "%d %s",
+ pjsua_var.nat_type,
+ type_name);
+ }
+
+ a = pjmedia_sdp_attr_create(pool, "X-nat", &value);
+
+ pjmedia_sdp_attr_add(&sdp->attr_count, sdp->attr, a);
+
+ }
+
+
+ /* Add bandwidth info in session level using bandwidth modifier "AS". */
+ if (tot_bandw_tias) {
+ unsigned bandw;
+ const pj_str_t STR_BANDW_MODIFIER_AS = { "AS", 2 };
+ pjmedia_sdp_bandw *b;
+
+ /* AS bandwidth = RTP bitrate + RTCP bitrate.
+ * RTP bitrate = payload bitrate (total TIAS) + overheads (~16kbps).
+ * RTCP bitrate = est. 5% of RTP bitrate.
+ * Note that AS bandwidth is in kbps.
+ */
+ bandw = tot_bandw_tias + 16000;
+ bandw += bandw * 5 / 100;
+ b = PJ_POOL_ALLOC_T(pool, pjmedia_sdp_bandw);
+ b->modifier = STR_BANDW_MODIFIER_AS;
+ b->value = bandw / 1000;
+ sdp->bandw[sdp->bandw_count++] = b;
+ }
+
+
+#if DISABLED_FOR_TICKET_1185 && defined(PJMEDIA_HAS_SRTP) && (PJMEDIA_HAS_SRTP != 0)
+ /* Check if SRTP is in optional mode and configured to use duplicated
+ * media, i.e: secured and unsecured version, in the SDP offer.
+ */
+ if (!rem_sdp &&
+ pjsua_var.acc[call->acc_id].cfg.use_srtp == PJMEDIA_SRTP_OPTIONAL &&
+ pjsua_var.acc[call->acc_id].cfg.srtp_optional_dup_offer)
+ {
+ unsigned i;
+
+ for (i = 0; i < sdp->media_count; ++i) {
+ pjmedia_sdp_media *m = sdp->media[i];
+
+ /* Check if this media is unsecured but has SDP "crypto"
+ * attribute.
+ */
+ if (pj_stricmp2(&m->desc.transport, "RTP/AVP") == 0 &&
+ pjmedia_sdp_media_find_attr2(m, "crypto", NULL) != NULL)
+ {
+ if (i == (unsigned)call->audio_idx &&
+ sdp_neg_state == PJMEDIA_SDP_NEG_STATE_DONE)
+ {
+ /* This is a session update, and peer has chosen the
+ * unsecured version, so let's make this unsecured too.
+ */
+ pjmedia_sdp_media_remove_all_attr(m, "crypto");
+ } else {
+ /* This is new offer, duplicate media so we'll have
+ * secured (with "RTP/SAVP" transport) and and unsecured
+ * versions.
+ */
+ pjmedia_sdp_media *new_m;
+
+ /* Duplicate this media and apply secured transport */
+ new_m = pjmedia_sdp_media_clone(pool, m);
+ pj_strdup2(pool, &new_m->desc.transport, "RTP/SAVP");
+
+ /* Remove the "crypto" attribute in the unsecured media */
+ pjmedia_sdp_media_remove_all_attr(m, "crypto");
+
+ /* Insert the new media before the unsecured media */
+ if (sdp->media_count < PJMEDIA_MAX_SDP_MEDIA) {
+ pj_array_insert(sdp->media, sizeof(new_m),
+ sdp->media_count, i, &new_m);
+ ++sdp->media_count;
+ ++i;
+ }
+ }
+ }
+ }
+ }
+#endif
+
+ call->rem_offerer = (rem_sdp != NULL);
+
+ /* Notify application */
+ if (pjsua_var.ua_cfg.cb.on_call_sdp_created) {
+ (*pjsua_var.ua_cfg.cb.on_call_sdp_created)(call_id, sdp,
+ pool, rem_sdp);
+ }
+
+ *p_sdp = sdp;
+ return PJ_SUCCESS;
+}
+
+
+static void stop_media_stream(pjsua_call *call, unsigned med_idx)
+{
+ pjsua_call_media *call_med = &call->media[med_idx];
+
+ /* Check if stream does not exist */
+ if (med_idx >= call->med_cnt)
+ return;
+
+ pj_log_push_indent();
+
+ if (call_med->type == PJMEDIA_TYPE_AUDIO) {
+ pjsua_aud_stop_stream(call_med);
+ }
+
+#if PJMEDIA_HAS_VIDEO
+ else if (call_med->type == PJMEDIA_TYPE_VIDEO) {
+ pjsua_vid_stop_stream(call_med);
+ }
+#endif
+
+ PJ_LOG(4,(THIS_FILE, "Media stream call%02d:%d is destroyed",
+ call->index, med_idx));
+ call_med->prev_state = call_med->state;
+ call_med->state = PJSUA_CALL_MEDIA_NONE;
+
+ /* Try to sync recent changes to provisional media */
+ if (med_idx < call->med_prov_cnt &&
+ call->media_prov[med_idx].tp == call_med->tp)
+ {
+ pjsua_call_media *prov_med = &call->media_prov[med_idx];
+
+ /* Media state */
+ prov_med->prev_state = call_med->prev_state;
+ prov_med->state = call_med->state;
+
+ /* RTP seq/ts */
+ prov_med->rtp_tx_seq_ts_set = call_med->rtp_tx_seq_ts_set;
+ prov_med->rtp_tx_seq = call_med->rtp_tx_seq;
+ prov_med->rtp_tx_ts = call_med->rtp_tx_ts;
+
+ /* Stream */
+ if (call_med->type == PJMEDIA_TYPE_AUDIO) {
+ prov_med->strm.a.conf_slot = call_med->strm.a.conf_slot;
+ prov_med->strm.a.stream = call_med->strm.a.stream;
+ }
+#if PJMEDIA_HAS_VIDEO
+ else if (call_med->type == PJMEDIA_TYPE_VIDEO) {
+ prov_med->strm.v.cap_win_id = call_med->strm.v.cap_win_id;
+ prov_med->strm.v.rdr_win_id = call_med->strm.v.rdr_win_id;
+ prov_med->strm.v.stream = call_med->strm.v.stream;
+ }
+#endif
+ }
+
+ pj_log_pop_indent();
+}
+
+static void stop_media_session(pjsua_call_id call_id)
+{
+ pjsua_call *call = &pjsua_var.calls[call_id];
+ unsigned mi;
+
+ for (mi=0; mi<call->med_cnt; ++mi) {
+ stop_media_stream(call, mi);
+ }
+}
+
+pj_status_t pjsua_media_channel_deinit(pjsua_call_id call_id)
+{
+ pjsua_call *call = &pjsua_var.calls[call_id];
+ unsigned mi;
+
+ for (mi=0; mi<call->med_cnt; ++mi) {
+ pjsua_call_media *call_med = &call->media[mi];
+
+ if (call_med->tp_st == PJSUA_MED_TP_CREATING) {
+ /* We will do the deinitialization after media transport
+ * creation is completed.
+ */
+ call->async_call.med_ch_deinit = PJ_TRUE;
+ return PJ_SUCCESS;
+ }
+ }
+
+ PJ_LOG(4,(THIS_FILE, "Call %d: deinitializing media..", call_id));
+ pj_log_push_indent();
+
+ stop_media_session(call_id);
+
+ /* Clean up media transports */
+ pjsua_media_prov_clean_up(call_id);
+ call->med_prov_cnt = 0;
+ for (mi=0; mi<call->med_cnt; ++mi) {
+ pjsua_call_media *call_med = &call->media[mi];
+
+ if (call_med->tp_st > PJSUA_MED_TP_IDLE) {
+ pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_IDLE);
+ pjmedia_transport_media_stop(call_med->tp);
+ }
+
+ if (call_med->tp) {
+ pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_NULL);
+ pjmedia_transport_close(call_med->tp);
+ call_med->tp = call_med->tp_orig = NULL;
+ }
+ call_med->tp_orig = NULL;
+ }
+
+ pj_log_pop_indent();
+
+ return PJ_SUCCESS;
+}
+
+
+/* Match codec fmtp. This will compare the values and the order. */
+static pj_bool_t match_codec_fmtp(const pjmedia_codec_fmtp *fmtp1,
+ const pjmedia_codec_fmtp *fmtp2)
+{
+ unsigned i;
+
+ if (fmtp1->cnt != fmtp2->cnt)
+ return PJ_FALSE;
+
+ for (i = 0; i < fmtp1->cnt; ++i) {
+ if (pj_stricmp(&fmtp1->param[i].name, &fmtp2->param[i].name))
+ return PJ_FALSE;
+ if (pj_stricmp(&fmtp1->param[i].val, &fmtp2->param[i].val))
+ return PJ_FALSE;
+ }
+
+ return PJ_TRUE;
+}
+
+#if PJSUA_MEDIA_HAS_PJMEDIA || PJSUA_THIRD_PARTY_STREAM_HAS_GET_INFO
+
+static pj_bool_t is_ice_running(pjmedia_transport *tp)
+{
+ pjmedia_transport_info tpinfo;
+ pjmedia_ice_transport_info *ice_info;
+
+ pjmedia_transport_info_init(&tpinfo);
+ pjmedia_transport_get_info(tp, &tpinfo);
+ ice_info = (pjmedia_ice_transport_info*)
+ pjmedia_transport_info_get_spc_info(&tpinfo,
+ PJMEDIA_TRANSPORT_TYPE_ICE);
+ return (ice_info && ice_info->sess_state == PJ_ICE_STRANS_STATE_RUNNING);
+}
+
+
+static pj_bool_t is_media_changed(const pjsua_call *call,
+ unsigned med_idx,
+ const pjsua_stream_info *new_si_)
+{
+ const pjsua_call_media *call_med = &call->media[med_idx];
+
+ /* Check for newly added media */
+ if (med_idx >= call->med_cnt)
+ return PJ_TRUE;
+
+ /* Compare media type */
+ if (call_med->type != new_si_->type)
+ return PJ_TRUE;
+
+ /* Audio update checks */
+ if (call_med->type == PJMEDIA_TYPE_AUDIO) {
+ pjmedia_stream_info the_old_si;
+ const pjmedia_stream_info *old_si = NULL;
+ const pjmedia_stream_info *new_si = &new_si_->info.aud;
+ const pjmedia_codec_info *old_ci = NULL;
+ const pjmedia_codec_info *new_ci = &new_si->fmt;
+ const pjmedia_codec_param *old_cp = NULL;
+ const pjmedia_codec_param *new_cp = new_si->param;
+
+ /* Compare media direction */
+ if (call_med->dir != new_si->dir)
+ return PJ_TRUE;
+
+ /* Get current active stream info */
+ if (call_med->strm.a.stream) {
+ pjmedia_stream_get_info(call_med->strm.a.stream, &the_old_si);
+ old_si = &the_old_si;
+ old_ci = &old_si->fmt;
+ old_cp = old_si->param;
+ } else {
+ /* The stream is inactive. */
+ return (new_si->dir != PJMEDIA_DIR_NONE);
+ }
+
+ /* Compare remote RTP address. If ICE is running, change in default
+ * address can happen after negotiation, this can be handled
+ * internally by ICE and does not need to cause media restart.
+ */
+ if (!is_ice_running(call_med->tp) &&
+ pj_sockaddr_cmp(&old_si->rem_addr, &new_si->rem_addr))
+ {
+ return PJ_TRUE;
+ }
+
+ /* Compare codec info */
+ if (pj_stricmp(&old_ci->encoding_name, &new_ci->encoding_name) ||
+ old_ci->clock_rate != new_ci->clock_rate ||
+ old_ci->channel_cnt != new_ci->channel_cnt ||
+ old_si->rx_pt != new_si->rx_pt ||
+ old_si->tx_pt != new_si->tx_pt ||
+ old_si->rx_event_pt != new_si->tx_event_pt ||
+ old_si->tx_event_pt != new_si->tx_event_pt)
+ {
+ return PJ_TRUE;
+ }
+
+ /* Compare codec param */
+ if (old_cp->setting.frm_per_pkt != new_cp->setting.frm_per_pkt ||
+ old_cp->setting.vad != new_cp->setting.vad ||
+ old_cp->setting.cng != new_cp->setting.cng ||
+ old_cp->setting.plc != new_cp->setting.plc ||
+ old_cp->setting.penh != new_cp->setting.penh ||
+ !match_codec_fmtp(&old_cp->setting.dec_fmtp,
+ &new_cp->setting.dec_fmtp) ||
+ !match_codec_fmtp(&old_cp->setting.enc_fmtp,
+ &new_cp->setting.enc_fmtp))
+ {
+ return PJ_TRUE;
+ }
+ }
+
+#if PJMEDIA_HAS_VIDEO
+ else if (call_med->type == PJMEDIA_TYPE_VIDEO) {
+ pjmedia_vid_stream_info the_old_si;
+ const pjmedia_vid_stream_info *old_si = NULL;
+ const pjmedia_vid_stream_info *new_si = &new_si_->info.vid;
+ const pjmedia_vid_codec_info *old_ci = NULL;
+ const pjmedia_vid_codec_info *new_ci = &new_si->codec_info;
+ const pjmedia_vid_codec_param *old_cp = NULL;
+ const pjmedia_vid_codec_param *new_cp = new_si->codec_param;
+
+ /* Compare media direction */
+ if (call_med->dir != new_si->dir)
+ return PJ_TRUE;
+
+ /* Get current active stream info */
+ if (call_med->strm.v.stream) {
+ pjmedia_vid_stream_get_info(call_med->strm.v.stream, &the_old_si);
+ old_si = &the_old_si;
+ old_ci = &old_si->codec_info;
+ old_cp = old_si->codec_param;
+ } else {
+ /* The stream is inactive. */
+ return (new_si->dir != PJMEDIA_DIR_NONE);
+ }
+
+ /* Compare remote RTP address. If ICE is running, change in default
+ * address can happen after negotiation, this can be handled
+ * internally by ICE and does not need to cause media restart.
+ */
+ if (!is_ice_running(call_med->tp) &&
+ pj_sockaddr_cmp(&old_si->rem_addr, &new_si->rem_addr))
+ {
+ return PJ_TRUE;
+ }
+
+ /* Compare codec info */
+ if (pj_stricmp(&old_ci->encoding_name, &new_ci->encoding_name) ||
+ old_si->rx_pt != new_si->rx_pt ||
+ old_si->tx_pt != new_si->tx_pt)
+ {
+ return PJ_TRUE;
+ }
+
+ /* Compare codec param */
+ if (/* old_cp->enc_mtu != new_cp->enc_mtu || */
+ pj_memcmp(&old_cp->enc_fmt.det, &new_cp->enc_fmt.det,
+ sizeof(pjmedia_video_format_detail)) ||
+ !match_codec_fmtp(&old_cp->dec_fmtp, &new_cp->dec_fmtp) ||
+ !match_codec_fmtp(&old_cp->enc_fmtp, &new_cp->enc_fmtp))
+ {
+ return PJ_TRUE;
+ }
+ }
+
+#endif
+
+ else {
+ /* Just return PJ_TRUE for other media type */
+ return PJ_TRUE;
+ }
+
+ return PJ_FALSE;
+}
+
+#else /* PJSUA_MEDIA_HAS_PJMEDIA || PJSUA_THIRD_PARTY_STREAM_HAS_GET_INFO */
+
+static pj_bool_t is_media_changed(const pjsua_call *call,
+ unsigned med_idx,
+ const pjsua_stream_info *new_si_)
+{
+ PJ_UNUSED_ARG(call);
+ PJ_UNUSED_ARG(med_idx);
+ PJ_UNUSED_ARG(new_si_);
+ /* Always assume that media has been changed */
+ return PJ_TRUE;
+}
+
+#endif /* PJSUA_MEDIA_HAS_PJMEDIA || PJSUA_THIRD_PARTY_STREAM_HAS_GET_INFO */
+
+
+pj_status_t pjsua_media_channel_update(pjsua_call_id call_id,
+ const pjmedia_sdp_session *local_sdp,
+ const pjmedia_sdp_session *remote_sdp)
+{
+ pjsua_call *call = &pjsua_var.calls[call_id];
+ pjsua_acc *acc = &pjsua_var.acc[call->acc_id];
+ pj_pool_t *tmp_pool = call->inv->pool_prov;
+ unsigned mi;
+ pj_bool_t got_media = PJ_FALSE;
+ pj_status_t status = PJ_SUCCESS;
+
+ const pj_str_t STR_AUDIO = { "audio", 5 };
+ const pj_str_t STR_VIDEO = { "video", 5 };
+ pj_uint8_t maudidx[PJSUA_MAX_CALL_MEDIA];
+ unsigned maudcnt = PJ_ARRAY_SIZE(maudidx);
+ unsigned mtotaudcnt = PJ_ARRAY_SIZE(maudidx);
+ pj_uint8_t mvididx[PJSUA_MAX_CALL_MEDIA];
+ unsigned mvidcnt = PJ_ARRAY_SIZE(mvididx);
+ unsigned mtotvidcnt = PJ_ARRAY_SIZE(mvididx);
+ pj_bool_t need_renego_sdp = PJ_FALSE;
+
+ if (pjsua_get_state() != PJSUA_STATE_RUNNING)
+ return PJ_EBUSY;
+
+ PJ_LOG(4,(THIS_FILE, "Call %d: updating media..", call_id));
+ pj_log_push_indent();
+
+ /* Destroy existing media session, if any. */
+ //stop_media_session(call->index);
+
+ /* Call media count must be at least equal to SDP media. Note that
+ * it may not be equal when remote removed any SDP media line.
+ */
+ pj_assert(call->med_prov_cnt >= local_sdp->media_count);
+
+ /* Reset audio_idx first */
+ call->audio_idx = -1;
+
+ /* Sort audio/video based on "quality" */
+ sort_media(local_sdp, &STR_AUDIO, acc->cfg.use_srtp,
+ maudidx, &maudcnt, &mtotaudcnt);
+#if PJMEDIA_HAS_VIDEO
+ sort_media(local_sdp, &STR_VIDEO, acc->cfg.use_srtp,
+ mvididx, &mvidcnt, &mtotvidcnt);
+#else
+ PJ_UNUSED_ARG(STR_VIDEO);
+ mvidcnt = mtotvidcnt = 0;
+#endif
+
+ /* Applying media count limitation. Note that in generating SDP answer,
+ * no media count limitation applied, as we didn't know yet which media
+ * would pass the SDP negotiation.
+ */
+ if (maudcnt > call->opt.aud_cnt || mvidcnt > call->opt.vid_cnt)
+ {
+ pjmedia_sdp_session *local_sdp2;
+
+ maudcnt = PJ_MIN(maudcnt, call->opt.aud_cnt);
+ mvidcnt = PJ_MIN(mvidcnt, call->opt.vid_cnt);
+ local_sdp2 = pjmedia_sdp_session_clone(tmp_pool, local_sdp);
+
+ for (mi=0; mi < local_sdp2->media_count; ++mi) {
+ pjmedia_sdp_media *m = local_sdp2->media[mi];
+
+ if (m->desc.port == 0 ||
+ pj_memchr(maudidx, mi, maudcnt*sizeof(maudidx[0])) ||
+ pj_memchr(mvididx, mi, mvidcnt*sizeof(mvididx[0])))
+ {
+ continue;
+ }
+
+ /* Deactivate this media */
+ pjmedia_sdp_media_deactivate(tmp_pool, m);
+ }
+
+ local_sdp = local_sdp2;
+ need_renego_sdp = PJ_TRUE;
+ }
+
+ /* Process each media stream */
+ for (mi=0; mi < call->med_prov_cnt; ++mi) {
+ pjsua_call_media *call_med = &call->media_prov[mi];
+ pj_bool_t media_changed = PJ_FALSE;
+
+ if (mi >= local_sdp->media_count ||
+ mi >= remote_sdp->media_count)
+ {
+ /* This may happen when remote removed any SDP media lines in
+ * its re-offer.
+ */
+
+ /* Stop stream */
+ stop_media_stream(call, mi);
+
+ /* Close the media transport */
+ if (call_med->tp) {
+ pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_NULL);
+ pjmedia_transport_close(call_med->tp);
+ call_med->tp = call_med->tp_orig = NULL;
+ }
+ continue;
+#if 0
+ /* Something is wrong */
+ PJ_LOG(1,(THIS_FILE, "Error updating media for call %d: "
+ "invalid media index %d in SDP", call_id, mi));
+ status = PJMEDIA_SDP_EINSDP;
+ goto on_error;
+#endif
+ }
+
+ /* Apply media update action */
+ if (call_med->type==PJMEDIA_TYPE_AUDIO) {
+ pjmedia_stream_info the_si, *si = &the_si;
+ pjsua_stream_info stream_info;
+
+ status = pjmedia_stream_info_from_sdp(
+ si, tmp_pool, pjsua_var.med_endpt,
+ local_sdp, remote_sdp, mi);
+ if (status != PJ_SUCCESS) {
+ PJ_PERROR(1,(THIS_FILE, status,
+ "pjmedia_stream_info_from_sdp() failed "
+ "for call_id %d media %d",
+ call_id, mi));
+ continue;
+ }
+
+ /* Codec parameter of stream info (si->param) can be NULL if
+ * the stream is rejected or disabled.
+ */
+ /* Override ptime, if this option is specified. */
+ if (pjsua_var.media_cfg.ptime != 0 && si->param) {
+ si->param->setting.frm_per_pkt = (pj_uint8_t)
+ (pjsua_var.media_cfg.ptime / si->param->info.frm_ptime);
+ if (si->param->setting.frm_per_pkt == 0)
+ si->param->setting.frm_per_pkt = 1;
+ }
+
+ /* Disable VAD, if this option is specified. */
+ if (pjsua_var.media_cfg.no_vad && si->param) {
+ si->param->setting.vad = 0;
+ }
+
+ /* Check if this media is changed */
+ stream_info.type = PJMEDIA_TYPE_AUDIO;
+ stream_info.info.aud = the_si;
+ if (pjsua_var.media_cfg.no_smart_media_update ||
+ is_media_changed(call, mi, &stream_info))
+ {
+ media_changed = PJ_TRUE;
+ /* Stop the media */
+ stop_media_stream(call, mi);
+ } else {
+ PJ_LOG(4,(THIS_FILE, "Call %d: stream #%d (audio) unchanged.",
+ call_id, mi));
+ }
+
+ /* Check if no media is active */
+ if (si->dir == PJMEDIA_DIR_NONE) {
+
+ /* Update call media state and direction */
+ call_med->state = PJSUA_CALL_MEDIA_NONE;
+ call_med->dir = PJMEDIA_DIR_NONE;
+
+ } else {
+ pjmedia_transport_info tp_info;
+ pjmedia_srtp_info *srtp_info;
+
+ /* Start/restart media transport based on info in SDP */
+ status = pjmedia_transport_media_start(call_med->tp,
+ tmp_pool, local_sdp,
+ remote_sdp, mi);
+ if (status != PJ_SUCCESS) {
+ PJ_PERROR(1,(THIS_FILE, status,
+ "pjmedia_transport_media_start() failed "
+ "for call_id %d media %d",
+ call_id, mi));
+ continue;
+ }
+
+ pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_RUNNING);
+
+ /* Get remote SRTP usage policy */
+ pjmedia_transport_info_init(&tp_info);
+ pjmedia_transport_get_info(call_med->tp, &tp_info);
+ srtp_info = (pjmedia_srtp_info*)
+ pjmedia_transport_info_get_spc_info(
+ &tp_info, PJMEDIA_TRANSPORT_TYPE_SRTP);
+ if (srtp_info) {
+ call_med->rem_srtp_use = srtp_info->peer_use;
+ }
+
+ /* Update audio channel */
+ if (media_changed) {
+ status = pjsua_aud_channel_update(call_med,
+ call->inv->pool, si,
+ local_sdp, remote_sdp);
+ if (status != PJ_SUCCESS) {
+ PJ_PERROR(1,(THIS_FILE, status,
+ "pjsua_aud_channel_update() failed "
+ "for call_id %d media %d",
+ call_id, mi));
+ continue;
+ }
+ }
+
+ /* Call media direction */
+ call_med->dir = si->dir;
+
+ /* Call media state */
+ if (call->local_hold)
+ call_med->state = PJSUA_CALL_MEDIA_LOCAL_HOLD;
+ else if (call_med->dir == PJMEDIA_DIR_DECODING)
+ call_med->state = PJSUA_CALL_MEDIA_REMOTE_HOLD;
+ else
+ call_med->state = PJSUA_CALL_MEDIA_ACTIVE;
+ }
+
+ /* Print info. */
+ if (status == PJ_SUCCESS) {
+ char info[80];
+ int info_len = 0;
+ int len;
+ const char *dir;
+
+ switch (si->dir) {
+ case PJMEDIA_DIR_NONE:
+ dir = "inactive";
+ break;
+ case PJMEDIA_DIR_ENCODING:
+ dir = "sendonly";
+ break;
+ case PJMEDIA_DIR_DECODING:
+ dir = "recvonly";
+ break;
+ case PJMEDIA_DIR_ENCODING_DECODING:
+ dir = "sendrecv";
+ break;
+ default:
+ dir = "unknown";
+ break;
+ }
+ len = pj_ansi_sprintf( info+info_len,
+ ", stream #%d: %.*s (%s)", mi,
+ (int)si->fmt.encoding_name.slen,
+ si->fmt.encoding_name.ptr,
+ dir);
+ if (len > 0)
+ info_len += len;
+ PJ_LOG(4,(THIS_FILE,"Audio updated%s", info));
+ }
+
+
+ if (call->audio_idx==-1 && status==PJ_SUCCESS &&
+ si->dir != PJMEDIA_DIR_NONE)
+ {
+ call->audio_idx = mi;
+ }
+
+#if defined(PJMEDIA_HAS_VIDEO) && (PJMEDIA_HAS_VIDEO != 0)
+ } else if (call_med->type==PJMEDIA_TYPE_VIDEO) {
+ pjmedia_vid_stream_info the_si, *si = &the_si;
+ pjsua_stream_info stream_info;
+
+ status = pjmedia_vid_stream_info_from_sdp(
+ si, tmp_pool, pjsua_var.med_endpt,
+ local_sdp, remote_sdp, mi);
+ if (status != PJ_SUCCESS) {
+ PJ_PERROR(1,(THIS_FILE, status,
+ "pjmedia_vid_stream_info_from_sdp() failed "
+ "for call_id %d media %d",
+ call_id, mi));
+ continue;
+ }
+
+ /* Check if this media is changed */
+ stream_info.type = PJMEDIA_TYPE_VIDEO;
+ stream_info.info.vid = the_si;
+ if (is_media_changed(call, mi, &stream_info)) {
+ media_changed = PJ_TRUE;
+ /* Stop the media */
+ stop_media_stream(call, mi);
+ } else {
+ PJ_LOG(4,(THIS_FILE, "Call %d: stream #%d (video) unchanged.",
+ call_id, mi));
+ }
+
+ /* Check if no media is active */
+ if (si->dir == PJMEDIA_DIR_NONE) {
+
+ /* Update call media state and direction */
+ call_med->state = PJSUA_CALL_MEDIA_NONE;
+ call_med->dir = PJMEDIA_DIR_NONE;
+
+ } else {
+ pjmedia_transport_info tp_info;
+ pjmedia_srtp_info *srtp_info;
+
+ /* Start/restart media transport */
+ status = pjmedia_transport_media_start(call_med->tp,
+ tmp_pool, local_sdp,
+ remote_sdp, mi);
+ if (status != PJ_SUCCESS) {
+ PJ_PERROR(1,(THIS_FILE, status,
+ "pjmedia_transport_media_start() failed "
+ "for call_id %d media %d",
+ call_id, mi));
+ continue;
+ }
+
+ pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_RUNNING);
+
+ /* Get remote SRTP usage policy */
+ pjmedia_transport_info_init(&tp_info);
+ pjmedia_transport_get_info(call_med->tp, &tp_info);
+ srtp_info = (pjmedia_srtp_info*)
+ pjmedia_transport_info_get_spc_info(
+ &tp_info, PJMEDIA_TRANSPORT_TYPE_SRTP);
+ if (srtp_info) {
+ call_med->rem_srtp_use = srtp_info->peer_use;
+ }
+
+ /* Update audio channel */
+ if (media_changed) {
+ status = pjsua_vid_channel_update(call_med,
+ call->inv->pool, si,
+ local_sdp, remote_sdp);
+ if (status != PJ_SUCCESS) {
+ PJ_PERROR(1,(THIS_FILE, status,
+ "pjsua_vid_channel_update() failed "
+ "for call_id %d media %d",
+ call_id, mi));
+ continue;
+ }
+ }
+
+ /* Call media direction */
+ call_med->dir = si->dir;
+
+ /* Call media state */
+ if (call->local_hold)
+ call_med->state = PJSUA_CALL_MEDIA_LOCAL_HOLD;
+ else if (call_med->dir == PJMEDIA_DIR_DECODING)
+ call_med->state = PJSUA_CALL_MEDIA_REMOTE_HOLD;
+ else
+ call_med->state = PJSUA_CALL_MEDIA_ACTIVE;
+ }
+
+ /* Print info. */
+ {
+ char info[80];
+ int info_len = 0;
+ int len;
+ const char *dir;
+
+ switch (si->dir) {
+ case PJMEDIA_DIR_NONE:
+ dir = "inactive";
+ break;
+ case PJMEDIA_DIR_ENCODING:
+ dir = "sendonly";
+ break;
+ case PJMEDIA_DIR_DECODING:
+ dir = "recvonly";
+ break;
+ case PJMEDIA_DIR_ENCODING_DECODING:
+ dir = "sendrecv";
+ break;
+ default:
+ dir = "unknown";
+ break;
+ }
+ len = pj_ansi_sprintf( info+info_len,
+ ", stream #%d: %.*s (%s)", mi,
+ (int)si->codec_info.encoding_name.slen,
+ si->codec_info.encoding_name.ptr,
+ dir);
+ if (len > 0)
+ info_len += len;
+ PJ_LOG(4,(THIS_FILE,"Video updated%s", info));
+ }
+
+#endif
+ } else {
+ status = PJMEDIA_EINVALIMEDIATYPE;
+ }
+
+ /* Close the transport of deactivated media, need this here as media
+ * can be deactivated by the SDP negotiation and the max media count
+ * (account) setting.
+ */
+ if (local_sdp->media[mi]->desc.port==0 && call_med->tp) {
+ pjsua_set_media_tp_state(call_med, PJSUA_MED_TP_NULL);
+ pjmedia_transport_close(call_med->tp);
+ call_med->tp = call_med->tp_orig = NULL;
+ }
+
+ if (status != PJ_SUCCESS) {
+ PJ_PERROR(1,(THIS_FILE, status, "Error updating media call%02d:%d",
+ call_id, mi));
+ } else {
+ got_media = PJ_TRUE;
+ }
+ }
+
+ /* Update call media from provisional media */
+ call->med_cnt = call->med_prov_cnt;
+ pj_memcpy(call->media, call->media_prov,
+ sizeof(call->media_prov[0]) * call->med_prov_cnt);
+
+ /* Perform SDP re-negotiation if needed. */
+ if (got_media && need_renego_sdp) {
+ pjmedia_sdp_neg *neg = call->inv->neg;
+
+ /* This should only happen when we are the answerer. */
+ PJ_ASSERT_RETURN(neg && !pjmedia_sdp_neg_was_answer_remote(neg),
+ PJMEDIA_SDPNEG_EINSTATE);
+
+ status = pjmedia_sdp_neg_set_remote_offer(tmp_pool, neg, remote_sdp);
+ if (status != PJ_SUCCESS)
+ goto on_error;
+
+ status = pjmedia_sdp_neg_set_local_answer(tmp_pool, neg, local_sdp);
+ if (status != PJ_SUCCESS)
+ goto on_error;
+
+ status = pjmedia_sdp_neg_negotiate(tmp_pool, neg, 0);
+ if (status != PJ_SUCCESS)
+ goto on_error;
+ }
+
+ pj_log_pop_indent();
+ return (got_media? PJ_SUCCESS : PJMEDIA_SDPNEG_ENOMEDIA);
+
+on_error:
+ pj_log_pop_indent();
+ return status;
+}
+
+/*****************************************************************************
+ * Codecs.
+ */
+
+/*
+ * Enum all supported codecs in the system.
+ */
+PJ_DEF(pj_status_t) pjsua_enum_codecs( pjsua_codec_info id[],
+ unsigned *p_count )
+{
+ pjmedia_codec_mgr *codec_mgr;
+ pjmedia_codec_info info[32];
+ unsigned i, count, prio[32];
+ pj_status_t status;
+
+ codec_mgr = pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt);
+ count = PJ_ARRAY_SIZE(info);
+ status = pjmedia_codec_mgr_enum_codecs( codec_mgr, &count, info, prio);
+ if (status != PJ_SUCCESS) {
+ *p_count = 0;
+ return status;
+ }
+
+ if (count > *p_count) count = *p_count;
+
+ for (i=0; i<count; ++i) {
+ pj_bzero(&id[i], sizeof(pjsua_codec_info));
+
+ pjmedia_codec_info_to_id(&info[i], id[i].buf_, sizeof(id[i].buf_));
+ id[i].codec_id = pj_str(id[i].buf_);
+ id[i].priority = (pj_uint8_t) prio[i];
+ }
+
+ *p_count = count;
+
+ return PJ_SUCCESS;
+}
+
+
+/*
+ * Change codec priority.
+ */
+PJ_DEF(pj_status_t) pjsua_codec_set_priority( const pj_str_t *codec_id,
+ pj_uint8_t priority )
+{
+ const pj_str_t all = { NULL, 0 };
+ pjmedia_codec_mgr *codec_mgr;
+
+ codec_mgr = pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt);
+
+ if (codec_id->slen==1 && *codec_id->ptr=='*')
+ codec_id = &all;
+
+ return pjmedia_codec_mgr_set_codec_priority(codec_mgr, codec_id,
+ priority);
+}
+
+
+/*
+ * Get codec parameters.
+ */
+PJ_DEF(pj_status_t) pjsua_codec_get_param( const pj_str_t *codec_id,
+ pjmedia_codec_param *param )
+{
+ const pj_str_t all = { NULL, 0 };
+ const pjmedia_codec_info *info;
+ pjmedia_codec_mgr *codec_mgr;
+ unsigned count = 1;
+ pj_status_t status;
+
+ codec_mgr = pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt);
+
+ if (codec_id->slen==1 && *codec_id->ptr=='*')
+ codec_id = &all;
+
+ status = pjmedia_codec_mgr_find_codecs_by_id(codec_mgr, codec_id,
+ &count, &info, NULL);
+ if (status != PJ_SUCCESS)
+ return status;
+
+ if (count != 1)
+ return (count > 1? PJ_ETOOMANY : PJ_ENOTFOUND);
+
+ status = pjmedia_codec_mgr_get_default_param( codec_mgr, info, param);
+ return status;
+}
+
+
+/*
+ * Set codec parameters.
+ */
+PJ_DEF(pj_status_t) pjsua_codec_set_param( const pj_str_t *codec_id,
+ const pjmedia_codec_param *param)
+{
+ const pjmedia_codec_info *info[2];
+ pjmedia_codec_mgr *codec_mgr;
+ unsigned count = 2;
+ pj_status_t status;
+
+ codec_mgr = pjmedia_endpt_get_codec_mgr(pjsua_var.med_endpt);
+
+ status = pjmedia_codec_mgr_find_codecs_by_id(codec_mgr, codec_id,
+ &count, info, NULL);
+ if (status != PJ_SUCCESS)
+ return status;
+
+ /* Codec ID should be specific, except for G.722.1 */
+ if (count > 1 &&
+ pj_strnicmp2(codec_id, "G7221/16", 8) != 0 &&
+ pj_strnicmp2(codec_id, "G7221/32", 8) != 0)
+ {
+ pj_assert(!"Codec ID is not specific");
+ return PJ_ETOOMANY;
+ }
+
+ status = pjmedia_codec_mgr_set_default_param(codec_mgr, info[0], param);
+ return status;
+}
+
+
+pj_status_t pjsua_media_apply_xml_control(pjsua_call_id call_id,
+ const pj_str_t *xml_st)
+{
+#if PJMEDIA_HAS_VIDEO
+ pjsua_call *call = &pjsua_var.calls[call_id];
+ const pj_str_t PICT_FAST_UPDATE = {"picture_fast_update", 19};
+
+ if (pj_strstr(xml_st, &PICT_FAST_UPDATE)) {
+ unsigned i;
+
+ PJ_LOG(4,(THIS_FILE, "Received keyframe request via SIP INFO"));
+
+ for (i = 0; i < call->med_cnt; ++i) {
+ pjsua_call_media *cm = &call->media[i];
+ if (cm->type != PJMEDIA_TYPE_VIDEO || !cm->strm.v.stream)
+ continue;
+
+ pjmedia_vid_stream_send_keyframe(cm->strm.v.stream);
+ }
+
+ return PJ_SUCCESS;
+ }
+#endif
+
+ /* Just to avoid compiler warning of unused var */
+ PJ_UNUSED_ARG(call_id);
+ PJ_UNUSED_ARG(xml_st);
+
+ return PJ_ENOTSUP;
+}
+