diff --git a/jni/pjproject-android/.svn/pristine/c2/c26eedb61e31b2c1a43e051ea1da92385fe8510c.svn-base b/jni/pjproject-android/.svn/pristine/c2/c26eedb61e31b2c1a43e051ea1da92385fe8510c.svn-base
new file mode 100644
index 0000000..acaa0e1
--- /dev/null
+++ b/jni/pjproject-android/.svn/pristine/c2/c26eedb61e31b2c1a43e051ea1da92385fe8510c.svn-base
@@ -0,0 +1,3058 @@
+/* $Id$ */
+/* 
+ * 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)
+{
+    unsigned i;
+
+    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();
+    }
+
+    /* 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);
+    }
+
+    /* 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;
+}
+