/plugins/janus_videocall.c

Large files files are truncated, but you can click here to view the full file

/*! \file   janus_videocall.c
 * \author Lorenzo Miniero <lorenzo@meetecho.com>
 * \copyright GNU General Public License v3
 * \brief  Janus VideoCall plugin
 * \details  This is a simple video call plugin for Janus, allowing two
 * WebRTC peers to call each other through the gateway. The idea is to
 * provide a similar service as the well known AppRTC demo (https://apprtc.appspot.com),
 * but with the media flowing through the gateway rather than being peer-to-peer.
 * 
 * The plugin provides a simple fake registration mechanism. A peer attaching
 * to the plugin needs to specify a username, which acts as a "phone number":
 * if the username is free, it is associated with the peer, which means
 * he/she can be "called" using that username by another peer. Peers can
 * either "call" another peer, by specifying their username, or wait for a call.
 * The approach used by this plugin is similar to the one employed by the
 * echo test one: all frames (RTP/RTCP) coming from one peer are relayed
 * to the other.
 * 
 * Just as in the janus_videocall.c plugin, there are knobs to control
 * whether audio and/or video should be muted or not, and if the bitrate
 * of the peer needs to be capped by means of REMB messages.
 * 
 * \section vcallapi Video Call API
 * 
 * All requests you can send in the Video Call API are asynchronous,
 * which means all responses (successes and errors) will be delivered
 * as events with the same transaction. 
 * 
 * The supported requests are \c list , \c register , \c call ,
 * \c accept , \c set and \c hangup . \c list allows you to get a list
 * of all the registered peers; \c register can be used to register
 * a username to call and be called; \c call is used to start a video
 * call with somebody through the plugin, while \c accept is used to
 * accept the call in case one is invited instead of inviting; \c set
 * can be used to configure some call-related settings (e.g., a cap on
 * the send bandwidth); finally, \c hangup can be used to terminate the
 * communication at any time, either to hangup an ongoing call or to
 * cancel/decline a call that hasn't started yet.
 * 
 * The \c list request has to be formatted as follows:
 * 
\verbatim
{
	"request" : "list"
}
\endverbatim
 *
 * A successful request will result in an array of peers to be returned:
 * 
\verbatim
{
	"videocall" : "event",
	"result" : {
		"list": [	// Array of peers
			"alice78",
			"bob51",
			// others
		]
	}
}
\endverbatim
 * 
 * An error instead (and the same applies to all other requests, so this
 * won't be repeated) would provide both an error code and a more verbose
 * description of the cause of the issue:
 * 
\verbatim
{
	"videocall" : "event",
	"error_code" : <numeric ID, check Macros below>,
	"error" : "<error description as a string>"
}
\endverbatim
 * 
 * To register a username to call and be called, the \c register request
 * can be used. This works on a "first come, first served" basis: there's
 * no authetication involved, you just specify the username you'd like
 * to use and, if free, it's assigned to you. The \c request has to be
 * formatted as follows:
 * 
\verbatim
{
	"request" : "register",
	"username" : "<desired unique username>"
}
\endverbatim
 * 
 * If successul, this will result in a \c registered event:
 * 
\verbatim
{
	"videocall" : "event",
	"result" : {
		"event" : "registered",
		"username" : "<same username, registered>"
	}
}
\endverbatim
 * 
 * Once you're registered, you can either start a new call or wait to
 * be called by someone else who knows your username. To start a new
 * call, the \c call request can be used: this request must be attached
 * to a JSEP offer containing the WebRTC-related info to setup a new
 * media session. A \c call request has to be formatted as follows:
 * 
\verbatim
{
	"request" : "call",
	"username" : "<username to call>"
}
\endverbatim
 * 
 * If successul, this will result in a \c calling event:
 * 
\verbatim
{
	"videocall" : "event",
	"result" : {
		"event" : "calling",
		"username" : "<same username, registered>"
	}
}
\endverbatim
 *
 * At the same time, the user being called will receive an
 * \c incomingcall event
 *  
\verbatim
{
	"videocall" : "event",
	"result" : {
		"event" : "incomingcall",
		"username" : "<your username>"
	}
}
\endverbatim
 * 
 * To accept the call, the \c accept request can be used. This request
 * must be attached to a JSEP answer containing the WebRTC-related
 * information to complete the actual PeerConnection setup. A \c accept
 * request has to be formatted as follows:
 * 
\verbatim
{
	"request" : "accept"
}
\endverbatim
 * 
 * If successul, both the caller and the callee will receive an
 * \c accepted event to notify them about the success of the signalling:
 * 
\verbatim
{
	"videocall" : "event",
	"result" : {
		"event" : "accepted",
		"username" : "<caller username>"
	}
}
\endverbatim
 *
 * At this point, the media-related settings of the call can be modified
 * on either side by means of a \c set request, which acts pretty much
 * as the one in the \ref echoapi . The \c set request has to be
 * formatted as follows. All the attributes (except \c request) are
 * optional, so any request can contain a subset of them:
 *
\verbatim
{
	"request" : "set",
	"audio" : true|false,
	"video" : true|false,
	"bitrate" : <numeric bitrate value>,
	"record" : true|false,
	"filename" : <base path/filename to use for the recording>
}
\endverbatim
 *
 * \c audio instructs the plugin to do or do not relay audio frames;
 * \c video does the same for video; \c bitrate caps the bandwidth to
 * force on the browser encoding side (e.g., 128000 for 128kbps);
 * \c record enables or disables the recording of this peer; in case
 * recording is enabled, \c filename allows to specify a base
 * path/filename to use for the files (-audio.mjr and -video.mjr are
 * automatically appended). Beware that enabling the recording only
 * records this user's contribution, and not the whole call: to record
 * both sides, you need to enable recording for both the peers in the
 * call.
 * 
 * A successful request will result in a \c set event:
 * 
\verbatim
{
	"videocall" : "event",
	"result" : {
		"event" : "set"
	}
}
\endverbatim
 * 
 * To decline an incoming call, cancel an attempt to call or simply
 * hangup an ongoing conversation, the \c hangup request can be used,
 * which has to be formatted as follows:
 * 
\verbatim
{
	"request" : "hangup"
}
\endverbatim
 *
 * Whatever the reason of a call being closed (e.g., a \c hangup request,
 * a PeerConnection being closed, or something else), both parties in
 * the communication will receive a \c hangup event:
 * 
\verbatim
{
	"videocall" : "event",
	"result" : {
		"event" : "hangup",
		"username" : "<username of who closed the communication>",
		"reason" : "<description of what happened>"
	}
}
\endverbatim
 * 
 * \ingroup plugins
 * \ref plugins
 */

#include "plugin.h"

#include <jansson.h>

#include "../debug.h"
#include "../apierror.h"
#include "../config.h"
#include "../mutex.h"
#include "../record.h"
#include "../rtp.h"
#include "../rtcp.h"
#include "../utils.h"


/* Plugin information */
#define JANUS_VIDEOCALL_VERSION			6
#define JANUS_VIDEOCALL_VERSION_STRING	"0.0.6"
#define JANUS_VIDEOCALL_DESCRIPTION		"This is a simple video call plugin for Janus, allowing two WebRTC peers to call each other through the gateway."
#define JANUS_VIDEOCALL_NAME			"JANUS VideoCall plugin"
#define JANUS_VIDEOCALL_AUTHOR			"Meetecho s.r.l."
#define JANUS_VIDEOCALL_PACKAGE			"janus.plugin.videocall"

/* Plugin methods */
janus_plugin *create(void);
int janus_videocall_init(janus_callbacks *callback, const char *config_path);
void janus_videocall_destroy(void);
int janus_videocall_get_api_compatibility(void);
int janus_videocall_get_version(void);
const char *janus_videocall_get_version_string(void);
const char *janus_videocall_get_description(void);
const char *janus_videocall_get_name(void);
const char *janus_videocall_get_author(void);
const char *janus_videocall_get_package(void);
void janus_videocall_create_session(janus_plugin_session *handle, int *error);
struct janus_plugin_result *janus_videocall_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep);
void janus_videocall_setup_media(janus_plugin_session *handle);
void janus_videocall_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len);
void janus_videocall_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len);
void janus_videocall_incoming_data(janus_plugin_session *handle, char *buf, int len);
void janus_videocall_slow_link(janus_plugin_session *handle, int uplink, int video);
void janus_videocall_hangup_media(janus_plugin_session *handle);
void janus_videocall_destroy_session(janus_plugin_session *handle, int *error);
json_t *janus_videocall_query_session(janus_plugin_session *handle);

/* Plugin setup */
static janus_plugin janus_videocall_plugin =
	JANUS_PLUGIN_INIT (
		.init = janus_videocall_init,
		.destroy = janus_videocall_destroy,

		.get_api_compatibility = janus_videocall_get_api_compatibility,
		.get_version = janus_videocall_get_version,
		.get_version_string = janus_videocall_get_version_string,
		.get_description = janus_videocall_get_description,
		.get_name = janus_videocall_get_name,
		.get_author = janus_videocall_get_author,
		.get_package = janus_videocall_get_package,
		
		.create_session = janus_videocall_create_session,
		.handle_message = janus_videocall_handle_message,
		.setup_media = janus_videocall_setup_media,
		.incoming_rtp = janus_videocall_incoming_rtp,
		.incoming_rtcp = janus_videocall_incoming_rtcp,
		.incoming_data = janus_videocall_incoming_data,
		.slow_link = janus_videocall_slow_link,
		.hangup_media = janus_videocall_hangup_media,
		.destroy_session = janus_videocall_destroy_session,
		.query_session = janus_videocall_query_session,
	);

/* Plugin creator */
janus_plugin *create(void) {
	JANUS_LOG(LOG_VERB, "%s created!\n", JANUS_VIDEOCALL_NAME);
	return &janus_videocall_plugin;
}

/* Parameter validation */
static struct janus_json_parameter request_parameters[] = {
	{"request", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
};
static struct janus_json_parameter username_parameters[] = {
	{"username", JSON_STRING, JANUS_JSON_PARAM_REQUIRED}
};
static struct janus_json_parameter set_parameters[] = {
	{"audio", JANUS_JSON_BOOL, 0},
	{"video", JANUS_JSON_BOOL, 0},
	{"bitrate", JSON_INTEGER, JANUS_JSON_PARAM_POSITIVE},
	{"record", JANUS_JSON_BOOL, 0},
	{"filename", JSON_STRING, 0}
};

/* Useful stuff */
static volatile gint initialized = 0, stopping = 0;
static gboolean notify_events = TRUE;
static janus_callbacks *gateway = NULL;
static GThread *handler_thread;
static GThread *watchdog;
static void *janus_videocall_handler(void *data);

typedef struct janus_videocall_message {
	janus_plugin_session *handle;
	char *transaction;
	json_t *message;
	json_t *jsep;
} janus_videocall_message;
static GAsyncQueue *messages = NULL;
static janus_videocall_message exit_message;

static void janus_videocall_message_free(janus_videocall_message *msg) {
	if(!msg || msg == &exit_message)
		return;

	msg->handle = NULL;

	g_free(msg->transaction);
	msg->transaction = NULL;
	if(msg->message)
		json_decref(msg->message);
	msg->message = NULL;
	if(msg->jsep)
		json_decref(msg->jsep);
	msg->jsep = NULL;

	g_free(msg);
}

typedef struct janus_videocall_session {
	janus_plugin_session *handle;
	gchar *username;
	gboolean has_audio;
	gboolean has_video;
	gboolean has_data;
	gboolean audio_active;
	gboolean video_active;
	uint32_t bitrate;
	guint16 slowlink_count;
	struct janus_videocall_session *peer;
	janus_rtp_switching_context context;
	uint32_t ssrc[3];		/* Only needed in case VP8 simulcasting is involved */
	int rtpmapid_extmap_id;	/* Only needed in case Firefox's RID-based simulcasting is involved */
	char *rid[3];			/* Only needed in case Firefox's RID-based simulcasting is involved */
	int substream;			/* Which simulcast substream we should forward back */
	int substream_target;	/* As above, but to handle transitions (e.g., wait for keyframe) */
	int templayer;			/* Which simulcast temporal layer we should forward back */
	int templayer_target;	/* As above, but to handle transitions (e.g., wait for keyframe) */
	gint64 last_relayed;	/* When we relayed the last packet (used to detect when substreams become unavailable) */
	janus_vp8_simulcast_context simulcast_context;
	janus_recorder *arc;	/* The Janus recorder instance for this user's audio, if enabled */
	janus_recorder *vrc;	/* The Janus recorder instance for this user's video, if enabled */
	janus_recorder *drc;	/* The Janus recorder instance for this user's data, if enabled */
	janus_mutex rec_mutex;	/* Mutex to protect the recorders from race conditions */
	volatile gint hangingup;
	gint64 destroyed;	/* Time at which this session was marked as destroyed */
} janus_videocall_session;
static GHashTable *sessions;
static GList *old_sessions;
static janus_mutex sessions_mutex = JANUS_MUTEX_INITIALIZER;


/* Error codes */
#define JANUS_VIDEOCALL_ERROR_UNKNOWN_ERROR			499
#define JANUS_VIDEOCALL_ERROR_NO_MESSAGE			470
#define JANUS_VIDEOCALL_ERROR_INVALID_JSON			471
#define JANUS_VIDEOCALL_ERROR_INVALID_REQUEST		472
#define JANUS_VIDEOCALL_ERROR_REGISTER_FIRST		473
#define JANUS_VIDEOCALL_ERROR_INVALID_ELEMENT		474
#define JANUS_VIDEOCALL_ERROR_MISSING_ELEMENT		475
#define JANUS_VIDEOCALL_ERROR_USERNAME_TAKEN		476
#define JANUS_VIDEOCALL_ERROR_ALREADY_REGISTERED	477
#define JANUS_VIDEOCALL_ERROR_NO_SUCH_USERNAME		478
#define JANUS_VIDEOCALL_ERROR_USE_ECHO_TEST			479
#define JANUS_VIDEOCALL_ERROR_ALREADY_IN_CALL		480
#define JANUS_VIDEOCALL_ERROR_NO_CALL				481
#define JANUS_VIDEOCALL_ERROR_MISSING_SDP			482


/* VideoCall watchdog/garbage collector (sort of) */
static void *janus_videocall_watchdog(void *data) {
	JANUS_LOG(LOG_INFO, "VideoCall watchdog started\n");
	gint64 now = 0;
	while(g_atomic_int_get(&initialized) && !g_atomic_int_get(&stopping)) {
		janus_mutex_lock(&sessions_mutex);
		/* Iterate on all the sessions */
		now = janus_get_monotonic_time();
		if(old_sessions != NULL) {
			GList *sl = old_sessions;
			JANUS_LOG(LOG_HUGE, "Checking %d old VideoCall sessions...\n", g_list_length(old_sessions));
			while(sl) {
				janus_videocall_session *session = (janus_videocall_session *)sl->data;
				if(!session) {
					sl = sl->next;
					continue;
				}
				if(now-session->destroyed >= 5*G_USEC_PER_SEC) {
					/* We're lazy and actually get rid of the stuff only after a few seconds */
					JANUS_LOG(LOG_VERB, "Freeing old VideoCall session\n");
					GList *rm = sl->next;
					old_sessions = g_list_delete_link(old_sessions, sl);
					sl = rm;
					session->handle = NULL;
					g_free(session);
					session = NULL;
					continue;
				}
				sl = sl->next;
			}
		}
		janus_mutex_unlock(&sessions_mutex);
		g_usleep(500000);
	}
	JANUS_LOG(LOG_INFO, "VideoCall watchdog stopped\n");
	return NULL;
}


/* Plugin implementation */
int janus_videocall_init(janus_callbacks *callback, const char *config_path) {
	if(g_atomic_int_get(&stopping)) {
		/* Still stopping from before */
		return -1;
	}
	if(callback == NULL || config_path == NULL) {
		/* Invalid arguments */
		return -1;
	}

	/* Read configuration */
	char filename[255];
	g_snprintf(filename, 255, "%s/%s.cfg", config_path, JANUS_VIDEOCALL_PACKAGE);
	JANUS_LOG(LOG_VERB, "Configuration file: %s\n", filename);
	janus_config *config = janus_config_parse(filename);
	if(config != NULL) {
		janus_config_print(config);
		janus_config_item *events = janus_config_get_item_drilldown(config, "general", "events");
		if(events != NULL && events->value != NULL)
			notify_events = janus_is_true(events->value);
		if(!notify_events && callback->events_is_enabled()) {
			JANUS_LOG(LOG_WARN, "Notification of events to handlers disabled for %s\n", JANUS_VIDEOCALL_NAME);
		}
	}
	janus_config_destroy(config);
	config = NULL;
	
	sessions = g_hash_table_new(g_str_hash, g_str_equal);
	messages = g_async_queue_new_full((GDestroyNotify) janus_videocall_message_free);
	/* This is the callback we'll need to invoke to contact the gateway */
	gateway = callback;

	g_atomic_int_set(&initialized, 1);

	GError *error = NULL;
	/* Start the sessions watchdog */
	watchdog = g_thread_try_new("videocall watchdog", &janus_videocall_watchdog, NULL, &error);
	if(error != NULL) {
		g_atomic_int_set(&initialized, 0);
		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the VideoCall watchdog thread...\n", error->code, error->message ? error->message : "??");
		return -1;
	}
	/* Launch the thread that will handle incoming messages */
	handler_thread = g_thread_try_new("videocall handler", janus_videocall_handler, NULL, &error);
	if(error != NULL) {
		g_atomic_int_set(&initialized, 0);
		JANUS_LOG(LOG_ERR, "Got error %d (%s) trying to launch the VideoCall handler thread...\n", error->code, error->message ? error->message : "??");
		return -1;
	}
	JANUS_LOG(LOG_INFO, "%s initialized!\n", JANUS_VIDEOCALL_NAME);
	return 0;
}

void janus_videocall_destroy(void) {
	if(!g_atomic_int_get(&initialized))
		return;
	g_atomic_int_set(&stopping, 1);

	g_async_queue_push(messages, &exit_message);
	if(handler_thread != NULL) {
		g_thread_join(handler_thread);
		handler_thread = NULL;
	}
	if(watchdog != NULL) {
		g_thread_join(watchdog);
		watchdog = NULL;
	}
	/* FIXME We should destroy the sessions cleanly */
	janus_mutex_lock(&sessions_mutex);
	g_hash_table_destroy(sessions);
	janus_mutex_unlock(&sessions_mutex);
	g_async_queue_unref(messages);
	messages = NULL;
	sessions = NULL;
	g_atomic_int_set(&initialized, 0);
	g_atomic_int_set(&stopping, 0);
	JANUS_LOG(LOG_INFO, "%s destroyed!\n", JANUS_VIDEOCALL_NAME);
}

int janus_videocall_get_api_compatibility(void) {
	/* Important! This is what your plugin MUST always return: don't lie here or bad things will happen */
	return JANUS_PLUGIN_API_VERSION;
}

int janus_videocall_get_version(void) {
	return JANUS_VIDEOCALL_VERSION;
}

const char *janus_videocall_get_version_string(void) {
	return JANUS_VIDEOCALL_VERSION_STRING;
}

const char *janus_videocall_get_description(void) {
	return JANUS_VIDEOCALL_DESCRIPTION;
}

const char *janus_videocall_get_name(void) {
	return JANUS_VIDEOCALL_NAME;
}

const char *janus_videocall_get_author(void) {
	return JANUS_VIDEOCALL_AUTHOR;
}

const char *janus_videocall_get_package(void) {
	return JANUS_VIDEOCALL_PACKAGE;
}

void janus_videocall_create_session(janus_plugin_session *handle, int *error) {
	if(g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized)) {
		*error = -1;
		return;
	}	
	janus_videocall_session *session = (janus_videocall_session *)g_malloc0(sizeof(janus_videocall_session));
	session->handle = handle;
	session->has_audio = FALSE;
	session->has_video = FALSE;
	session->has_data = FALSE;
	session->audio_active = TRUE;
	session->video_active = TRUE;
	session->bitrate = 0;	/* No limit */
	session->peer = NULL;
	session->username = NULL;
	janus_rtp_switching_context_reset(&session->context);
	session->ssrc[0] = 0;
	session->ssrc[1] = 0;
	session->ssrc[2] = 0;
	session->substream = -1;
	session->substream_target = 0;
	session->templayer = -1;
	session->templayer_target = 0;
	session->last_relayed = 0;
	janus_vp8_simulcast_context_reset(&session->simulcast_context);
	janus_mutex_init(&session->rec_mutex);
	session->destroyed = 0;
	g_atomic_int_set(&session->hangingup, 0);
	handle->plugin_handle = session;

	return;
}

void janus_videocall_destroy_session(janus_plugin_session *handle, int *error) {
	if(g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized)) {
		*error = -1;
		return;
	}
	janus_videocall_session *session = (janus_videocall_session *)handle->plugin_handle; 
	if(!session) {
		JANUS_LOG(LOG_ERR, "No VideoCall session associated with this handle...\n");
		*error = -2;
		return;
	}
	janus_mutex_lock(&sessions_mutex);
	if(!session->destroyed) {
		JANUS_LOG(LOG_VERB, "Removing VideoCall user %s session...\n", session->username ? session->username : "'unknown'");
		janus_videocall_hangup_media(handle);
		session->destroyed = janus_get_monotonic_time();
		if(session->username != NULL) {
			int res = g_hash_table_remove(sessions, (gpointer)session->username);
			JANUS_LOG(LOG_VERB, "  -- Removed: %d\n", res);
		}
		/* Cleaning up and removing the session is done in a lazy way */
		old_sessions = g_list_append(old_sessions, session);
	}
	janus_mutex_unlock(&sessions_mutex);
	return;
}

json_t *janus_videocall_query_session(janus_plugin_session *handle) {
	if(g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized)) {
		return NULL;
	}	
	janus_videocall_session *session = (janus_videocall_session *)handle->plugin_handle;
	if(!session) {
		JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
		return NULL;
	}
	/* Provide some generic info, e.g., if we're in a call and with whom */
	json_t *info = json_object();
	json_object_set_new(info, "state", json_string(session->peer ? "incall" : "idle"));
	json_object_set_new(info, "username", session->username ? json_string(session->username) : NULL);
	if(session->peer) {
		json_object_set_new(info, "peer", session->peer->username ? json_string(session->peer->username) : NULL);
		json_object_set_new(info, "audio_active", session->audio_active ? json_true() : json_false());
		json_object_set_new(info, "video_active", session->video_active ? json_true() : json_false());
		json_object_set_new(info, "bitrate", json_integer(session->bitrate));
		json_object_set_new(info, "slowlink_count", json_integer(session->slowlink_count));
	}
	if(session->ssrc[0] != 0) {
		json_object_set_new(info, "simulcast", json_true());
	}
	if(session->peer && session->peer->ssrc[0] != 0) {
		json_object_set_new(info, "simulcast-peer", json_true());
		json_object_set_new(info, "substream", json_integer(session->substream));
		json_object_set_new(info, "substream-target", json_integer(session->substream_target));
		json_object_set_new(info, "temporal-layer", json_integer(session->templayer));
		json_object_set_new(info, "temporal-layer-target", json_integer(session->templayer_target));
	}
	if(session->arc || session->vrc || session->drc) {
		json_t *recording = json_object();
		if(session->arc && session->arc->filename)
			json_object_set_new(recording, "audio", json_string(session->arc->filename));
		if(session->vrc && session->vrc->filename)
			json_object_set_new(recording, "video", json_string(session->vrc->filename));
		if(session->drc && session->drc->filename)
			json_object_set_new(recording, "data", json_string(session->drc->filename));
		json_object_set_new(info, "recording", recording);
	}
	json_object_set_new(info, "destroyed", json_integer(session->destroyed));
	return info;
}

struct janus_plugin_result *janus_videocall_handle_message(janus_plugin_session *handle, char *transaction, json_t *message, json_t *jsep) {
	if(g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
		return janus_plugin_result_new(JANUS_PLUGIN_ERROR, g_atomic_int_get(&stopping) ? "Shutting down" : "Plugin not initialized", NULL);
	janus_videocall_message *msg = g_malloc0(sizeof(janus_videocall_message));
	msg->handle = handle;
	msg->transaction = transaction;
	msg->message = message;
	msg->jsep = jsep;
	g_async_queue_push(messages, msg);

	/* All the requests to this plugin are handled asynchronously */
	return janus_plugin_result_new(JANUS_PLUGIN_OK_WAIT, NULL, NULL);
}

void janus_videocall_setup_media(janus_plugin_session *handle) {
	JANUS_LOG(LOG_INFO, "WebRTC media is now available\n");
	if(g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
		return;
	janus_videocall_session *session = (janus_videocall_session *)handle->plugin_handle;	
	if(!session) {
		JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
		return;
	}
	if(session->destroyed)
		return;
	g_atomic_int_set(&session->hangingup, 0);
	/* We really don't care, as we only relay RTP/RTCP we get in the first place anyway */
}

void janus_videocall_incoming_rtp(janus_plugin_session *handle, int video, char *buf, int len) {
	if(handle == NULL || handle->stopped || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
		return;
	if(gateway) {
		/* Honour the audio/video active flags */
		janus_videocall_session *session = (janus_videocall_session *)handle->plugin_handle;	
		if(!session) {
			JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
			return;
		}
		if(!session->peer) {
			JANUS_LOG(LOG_ERR, "Session has no peer...\n");
			return;
		}
		if(session->destroyed || session->peer->destroyed)
			return;
		if(video && session->video_active && session->rtpmapid_extmap_id != -1) {
			/* FIXME Just a way to debug Firefox simulcasting */
			janus_rtp_header *header = (janus_rtp_header *)buf;
			uint32_t seq_number = ntohs(header->seq_number);
			uint32_t timestamp = ntohl(header->timestamp);
			uint32_t ssrc = ntohl(header->ssrc);
			char sdes_item[16];
			if(janus_rtp_header_extension_parse_rtp_stream_id(buf, len, session->rtpmapid_extmap_id, sdes_item, sizeof(sdes_item)) == 0) {
				JANUS_LOG(LOG_DBG, "%"SCNu32"/%"SCNu16"/%"SCNu32"/%d: RTP stream ID extension: %s\n",
					ssrc, seq_number, timestamp, header->padding, sdes_item);
			}
		}
		if(video && session->video_active && session->ssrc[0] != 0) {
			/* Handle simulcast: don't relay if it's not the SSRC we wanted to handle */
			janus_rtp_header *header = (janus_rtp_header *)buf;
			uint32_t seq_number = ntohs(header->seq_number);
			uint32_t timestamp = ntohl(header->timestamp);
			uint32_t ssrc = ntohl(header->ssrc);
			/* Access the packet payload */
			int plen = 0;
			char *payload = janus_rtp_payload(buf, len, &plen);
			if(payload == NULL)
				return;
			gboolean switched = FALSE;
			if(session->peer->substream != session->peer->substream_target) {
				/* There has been a change: let's wait for a keyframe on the target */
				int step = (session->peer->substream < 1 && session->peer->substream_target == 2);
				if((ssrc == session->ssrc[session->peer->substream_target]) || (step && ssrc == session->ssrc[step])) {
					//~ if(janus_vp8_is_keyframe(payload, plen)) {
						uint32_t ssrc_old = 0;
						if(session->peer->substream != -1)
							ssrc_old = session->ssrc[session->peer->substream];
						JANUS_LOG(LOG_VERB, "Received keyframe on SSRC %"SCNu32", switching (was %"SCNu32")\n", ssrc, ssrc_old);
						session->peer->substream = (ssrc == session->ssrc[session->peer->substream_target] ? session->peer->substream_target : step);
						switched = TRUE;
						/* Notify the peer */
						json_t *event = json_object();
						json_object_set_new(event, "videocall", json_string("event"));
						json_t *result = json_object();
						json_object_set_new(result, "event", json_string("simulcast"));
						json_object_set_new(result, "substream", json_integer(session->peer->substream));
						json_object_set_new(event, "result", result);
						gateway->push_event(session->peer->handle, &janus_videocall_plugin, NULL, event, NULL);
						json_decref(event);
					//~ } else {
						//~ JANUS_LOG(LOG_WARN, "Not a keyframe on SSRC %"SCNu32" yet, waiting before switching\n", ssrc);
					//~ }
				}
			}
			/* If we haven't received our desired substream yet, let's drop temporarily */
			if(session->last_relayed == 0) {
				/* Let's start slow */
				session->last_relayed = janus_get_monotonic_time();
			} else {
				/* Check if 250ms went by with no packet relayed */
				gint64 now = janus_get_monotonic_time();
				if(now-session->last_relayed >= 250000) {
					session->last_relayed = now;
					int substream = session->peer->substream-1;
					if(substream < 0)
						substream = 0;
					if(session->peer->substream != substream) {
						JANUS_LOG(LOG_WARN, "No packet received on substream %d for a while, falling back to %d\n",
							session->peer->substream, substream);
						session->peer->substream = substream;
						/* Send a PLI to the user */
						JANUS_LOG(LOG_VERB, "Just (re-)enabled video, sending a PLI to recover it\n");
						char rtcpbuf[12];
						memset(rtcpbuf, 0, 12);
						janus_rtcp_pli((char *)&rtcpbuf, 12);
						gateway->relay_rtcp(handle, 1, rtcpbuf, 12);
						/* Notify the peer */
						json_t *event = json_object();
						json_object_set_new(event, "videocall", json_string("event"));
						json_t *result = json_object();
						json_object_set_new(result, "event", json_string("simulcast"));
						json_object_set_new(result, "substream", json_integer(session->peer->substream));
						json_object_set_new(event, "result", result);
						gateway->push_event(session->peer->handle, &janus_videocall_plugin, NULL, event, NULL);
						json_decref(event);
					}
				}
			}
			/* Do we need to drop this? */
			if(ssrc != session->ssrc[session->peer->substream]) {
				JANUS_LOG(LOG_HUGE, "Dropping packet (it's from SSRC %"SCNu32", but we're only relaying to the peer the SSRC %"SCNu32" now\n",
					ssrc, session->ssrc[session->peer->substream]);
				return;
			}
			session->last_relayed = janus_get_monotonic_time();
			/* Check if there's any temporal scalability to take into account */
			uint16_t picid = 0;
			uint8_t tlzi = 0;
			uint8_t tid = 0;
			uint8_t ybit = 0;
			uint8_t keyidx = 0;
			if(janus_vp8_parse_descriptor(payload, plen, &picid, &tlzi, &tid, &ybit, &keyidx) == 0) {
				//~ JANUS_LOG(LOG_WARN, "%"SCNu16", %u, %u, %u, %u\n", picid, tlzi, tid, ybit, keyidx);
				if(session->peer->templayer != session->peer->templayer_target) {
					/* FIXME We should be smarter in deciding when to switch */
					session->peer->templayer = session->peer->templayer_target;
					/* Notify the peer */
					json_t *event = json_object();
						json_object_set_new(event, "videocall", json_string("event"));
						json_t *result = json_object();
						json_object_set_new(result, "event", json_string("simulcast"));
						json_object_set_new(result, "temporal", json_integer(session->peer->templayer));
						json_object_set_new(event, "result", result);
					gateway->push_event(session->peer->handle, &janus_videocall_plugin, NULL, event, NULL);
					json_decref(event);
				}
				if(tid > session->peer->templayer) {
					JANUS_LOG(LOG_HUGE, "Dropping packet (it's temporal layer %d, but we're capping at %d)\n",
						tid, session->peer->templayer);
					/* We increase the base sequence number, or there will be gaps when delivering later */
					session->peer->context.v_base_seq++;
					return;
				}
			}
			/* If we got here, update the RTP header and send the packet */
			janus_rtp_header_update(header, &session->peer->context, TRUE, 4500);
			janus_vp8_simulcast_descriptor_update(payload, plen, &session->peer->simulcast_context, switched);
			/* Save the frame if we're recording */
			janus_recorder_save_frame(session->vrc, buf, len);
			/* Send the frame back */
			gateway->relay_rtp(session->peer->handle, video, buf, len);
			/* Restore header or core statistics will be messed up */
			header->timestamp = htonl(timestamp);
			header->seq_number = htons(seq_number);
		} else {
			if((!video && session->audio_active) || (video && session->video_active)) {
				/* Save the frame if we're recording */
				janus_recorder_save_frame(video ? session->vrc : session->arc, buf, len);
				/* Forward the packet to the peer */
				gateway->relay_rtp(session->peer->handle, video, buf, len);
			}
		}
	}
}

void janus_videocall_incoming_rtcp(janus_plugin_session *handle, int video, char *buf, int len) {
	if(handle == NULL || handle->stopped || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
		return;
	if(gateway) {
		janus_videocall_session *session = (janus_videocall_session *)handle->plugin_handle;	
		if(!session) {
			JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
			return;
		}
		if(!session->peer) {
			JANUS_LOG(LOG_ERR, "Session has no peer...\n");
			return;
		}
		if(session->destroyed || session->peer->destroyed)
			return;
		guint32 bitrate = janus_rtcp_get_remb(buf, len);
		if(bitrate > 0) {
			/* If a REMB arrived, make sure we cap it to our configuration, and send it as a video RTCP */
			if(session->bitrate > 0)
				janus_rtcp_cap_remb(buf, len, session->bitrate);
			gateway->relay_rtcp(session->peer->handle, 1, buf, len);
			return;
		}
		gateway->relay_rtcp(session->peer->handle, video, buf, len);
	}
}

void janus_videocall_incoming_data(janus_plugin_session *handle, char *buf, int len) {
	if(handle == NULL || handle->stopped || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
		return;
	if(gateway) {
		janus_videocall_session *session = (janus_videocall_session *)handle->plugin_handle;	
		if(!session) {
			JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
			return;
		}
		if(!session->peer) {
			JANUS_LOG(LOG_ERR, "Session has no peer...\n");
			return;
		}
		if(session->destroyed || session->peer->destroyed)
			return;
		if(buf == NULL || len <= 0)
			return;
		char *text = g_malloc0(len+1);
		memcpy(text, buf, len);
		*(text+len) = '\0';
		JANUS_LOG(LOG_VERB, "Got a DataChannel message (%zu bytes) to forward: %s\n", strlen(text), text);
		gateway->relay_data(session->peer->handle, text, strlen(text));
		g_free(text);
	}
}

void janus_videocall_slow_link(janus_plugin_session *handle, int uplink, int video) {
	/* The core is informing us that our peer got or sent too many NACKs, are we pushing media too hard? */
	if(handle == NULL || handle->stopped || g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
		return;
	janus_videocall_session *session = (janus_videocall_session *)handle->plugin_handle;	
	if(!session) {
		JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
		return;
	}
	if(session->destroyed)
		return;
	session->slowlink_count++;
	if(uplink && !video && !session->audio_active) {
		/* We're not relaying audio and the peer is expecting it, so NACKs are normal */
		JANUS_LOG(LOG_VERB, "Getting a lot of NACKs (slow uplink) for audio, but that's expected, a configure disabled the audio forwarding\n");
	} else if(uplink && video && !session->video_active) {
		/* We're not relaying video and the peer is expecting it, so NACKs are normal */
		JANUS_LOG(LOG_VERB, "Getting a lot of NACKs (slow uplink) for video, but that's expected, a configure disabled the video forwarding\n");
	} else {
		/* Slow uplink or downlink, maybe we set the bitrate cap too high? */
		if(video) {
			/* Halve the bitrate, but don't go too low... */
			if(!uplink) {
				/* Downlink issue, user has trouble sending, halve this user's bitrate cap */
				session->bitrate = session->bitrate > 0 ? session->bitrate : 512*1024;
				session->bitrate = session->bitrate/2;
				if(session->bitrate < 64*1024)
					session->bitrate = 64*1024;
			} else {
				/* Uplink issue, user has trouble receiving, halve this user's peer's bitrate cap */
				if(session->peer == NULL || session->peer->handle == NULL)
					return;	/* Nothing to do */
				session->peer->bitrate = session->peer->bitrate > 0 ? session->peer->bitrate : 512*1024;
				session->peer->bitrate = session->peer->bitrate/2;
				if(session->peer->bitrate < 64*1024)
					session->peer->bitrate = 64*1024;
			}
			JANUS_LOG(LOG_WARN, "Getting a lot of NACKs (slow %s) for %s, forcing a lower REMB: %"SCNu32"\n",
				uplink ? "uplink" : "downlink", video ? "video" : "audio", uplink ? session->peer->bitrate : session->bitrate);
			/* ... and send a new REMB back */
			char rtcpbuf[24];
			janus_rtcp_remb((char *)(&rtcpbuf), 24, uplink ? session->peer->bitrate : session->bitrate);
			gateway->relay_rtcp(uplink ? session->peer->handle : handle, 1, rtcpbuf, 24);
			/* As a last thing, notify the affected user about this */
			json_t *event = json_object();
			json_object_set_new(event, "videocall", json_string("event"));
			json_t *result = json_object();
			json_object_set_new(result, "status", json_string("slow_link"));
			json_object_set_new(result, "bitrate", json_integer(uplink ? session->peer->bitrate : session->bitrate));
			json_object_set_new(event, "result", result);
			gateway->push_event(uplink ? session->peer->handle : handle, &janus_videocall_plugin, NULL, event, NULL);
			json_decref(event);
		}
	}
}

void janus_videocall_hangup_media(janus_plugin_session *handle) {
	JANUS_LOG(LOG_INFO, "No WebRTC media anymore\n");
	if(g_atomic_int_get(&stopping) || !g_atomic_int_get(&initialized))
		return;
	janus_videocall_session *session = (janus_videocall_session *)handle->plugin_handle;	
	if(!session) {
		JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
		return;
	}
	if(session->destroyed)
		return;
	if(g_atomic_int_add(&session->hangingup, 1))
		return;
	/* Get rid of the recorders, if available */
	janus_mutex_lock(&session->rec_mutex);
	if(session->arc) {
		janus_recorder_close(session->arc);
		JANUS_LOG(LOG_INFO, "Closed audio recording %s\n", session->arc->filename ? session->arc->filename : "??");
		janus_recorder_free(session->arc);
	}
	session->arc = NULL;
	if(session->vrc) {
		janus_recorder_close(session->vrc);
		JANUS_LOG(LOG_INFO, "Closed video recording %s\n", session->vrc->filename ? session->vrc->filename : "??");
		janus_recorder_free(session->vrc);
	}
	session->vrc = NULL;
	if(session->drc) {
		janus_recorder_close(session->drc);
		JANUS_LOG(LOG_INFO, "Closed data recording %s\n", session->drc->filename ? session->drc->filename : "??");
		janus_recorder_free(session->drc);
	}
	session->drc = NULL;
	janus_mutex_unlock(&session->rec_mutex);
	if(session->peer) {
		/* Send event to our peer too */
		json_t *call = json_object();
		json_object_set_new(call, "videocall", json_string("event"));
		json_t *calling = json_object();
		json_object_set_new(calling, "event", json_string("hangup"));
		json_object_set_new(calling, "username", json_string(session->username));
		json_object_set_new(calling, "reason", json_string("Remote WebRTC hangup"));
		json_object_set_new(call, "result", calling);
		gateway->close_pc(session->peer->handle);
		int ret = gateway->push_event(session->peer->handle, &janus_videocall_plugin, NULL, call, NULL);
		JANUS_LOG(LOG_VERB, "  >> Pushing event to peer: %d (%s)\n", ret, janus_get_api_error(ret));
		json_decref(call);
		/* Also notify event handlers */
		if(notify_events && gateway->events_is_enabled()) {
			json_t *info = json_object();
			json_object_set_new(info, "event", json_string("hangup"));
			json_object_set_new(info, "reason", json_string("Remote WebRTC hangup"));
			gateway->notify_event(&janus_videocall_plugin, session->peer->handle, info);
		}
	}
	session->peer = NULL;
	/* Reset controls */
	session->has_audio = FALSE;
	session->has_video = FALSE;
	session->has_data = FALSE;
	session->audio_active = TRUE;
	session->video_active = TRUE;
	session->bitrate = 0;
	janus_rtp_switching_context_reset(&session->context);
}

/* Thread to handle incoming messages */
static void *janus_videocall_handler(void *data) {
	JANUS_LOG(LOG_VERB, "Joining VideoCall handler thread\n");
	janus_videocall_message *msg = NULL;
	int error_code = 0;
	char error_cause[512];
	json_t *root = NULL;
	while(g_atomic_int_get(&initialized) && !g_atomic_int_get(&stopping)) {
		msg = g_async_queue_pop(messages);
		if(msg == NULL)
			continue;
		if(msg == &exit_message)
			break;
		if(msg->handle == NULL) {
			janus_videocall_message_free(msg);
			continue;
		}
		janus_mutex_lock(&sessions_mutex);
		janus_videocall_session *session = (janus_videocall_session *)msg->handle->plugin_handle;
		if(!session) {
			JANUS_LOG(LOG_ERR, "No session associated with this handle...\n");
			janus_mutex_unlock(&sessions_mutex);
			janus_videocall_message_free(msg);
			continue;
		}
		if(session->destroyed) {
			janus_mutex_unlock(&sessions_mutex);
			janus_videocall_message_free(msg);
			continue;
		}
		janus_mutex_unlock(&sessions_mutex);
		/* Handle request */
		error_code = 0;
		root = msg->message;
		if(msg->message == NULL) {
			JANUS_LOG(LOG_ERR, "No message??\n");
			error_code = JANUS_VIDEOCALL_ERROR_NO_MESSAGE;
			g_snprintf(error_cause, 512, "%s", "No message??");
			goto error;
		}
		if(!json_is_object(root)) {
			JANUS_LOG(LOG_ERR, "JSON error: not an object\n");
			error_code = JANUS_VIDEOCALL_ERROR_INVALID_JSON;
			g_snprintf(error_cause, 512, "JSON error: not an object");
			goto error;
		}
		JANUS_VALIDATE_JSON_OBJECT(root, request_parameters,
			error_code, error_cause, TRUE,
			JANUS_VIDEOCALL_ERROR_MISSING_ELEMENT, JANUS_VIDEOCALL_ERROR_INVALID_ELEMENT);
		if(error_code != 0)
			goto error;
		const char *msg_sdp_type = json_string_value(json_object_get(msg->jsep, "type"));
		const char *msg_sdp = json_string_value(json_object_get(msg->jsep, "sdp"));
		json_t *request = json_object_get(root, "request");
		const char *request_text = json_string_value(request);
		json_t *result = NULL;
		if(!strcasecmp(request_text, "list")) {
			result = json_object();
			json_t *list = json_array();
			JANUS_LOG(LOG_VERB, "Request for the list of peers\n");
			/* Return a list of all available mountpoints */
			janus_mutex_lock(&sessions_mutex);
			GHashTableIter iter;
			gpointer value;
			g_hash_table_iter_init(&iter, sessions);
			while (g_hash_table_iter_next(&iter, NULL, &value)) {
				janus_videocall_session *user = value;
				if(user != NULL && user->username != NULL)
					json_array_append_new(list, json_string(user->username));
			}
			json_object_set_new(result, "list", list);
			janus_mutex_unlock(&sessions_mutex);
		} else if(!strcasecmp(request_text, "register")) {
			/* Map this handle to a username */
			if(session->username != NULL) {
				JANUS_LOG(LOG_ERR, "Already registered (%s)\n", session->username);
				error_code = JANUS_VIDEOCALL_ERROR_ALREADY_REGISTERED;
				g_snprintf(error_cause, 512, "Already registered (%s)", session->username);
				goto error;
			}
			JANUS_VALIDATE_JSON_OBJECT(root, username_parameters,
				error_code, error_cause, TRUE,
				JANUS_VIDEOCALL_ERROR_MISSING_ELEMENT, JANUS_VIDEOCALL_ERROR_INVALID_ELEMENT);
			if(error_code != 0)
				goto error;
			json_t *username = json_object_get(root, "username");
			const char *username_text = json_string_value(username);
			janus_mutex_lock(&sessions_mutex);
			if(g_hash_table_lookup(sessions, username_text) != NULL) {
				janus_mutex_unlock(&sessions_mutex);
				JANUS_LOG(LOG_ERR, "Username '%s' already taken\n", username_text);
				error_code = JANUS_VIDEOCALL_ERROR_USERNAME_TAKEN;
				g_snprintf(error_cause, 512, "Username '%s' already taken", username_text);
				goto error;
			}
			janus_mutex_unlock(&sessions_mutex);
			session->username = g_strdup(username_text);
			janus_mutex_lock(&sessions_mutex);
			g_hash_table_insert(sessions, (gpointer)session->username, session);
			janus_mutex_unlock(&sessions_mutex);
			result = json_object();
			json_object_set_new(result, "event", json_string("registered"));
			json_object_set_new(result, "username", json_string(username_text));
			/* Also notify event handlers */
			if(notify_events && gateway->events_is_enabled()) {
				json_t *info = json_object();
				json_object_set_new(info, "event", json_string("registered"));
				json_object_set_new(info, "username", json_string(username_text));
				gateway->notify_event(&janus_videocall_plugin, session->handle, info);
			}
		} else if(!strcasecmp(request_text, "call")) {
			/* Call another peer */
			if(session->username == NULL) {
				JANUS_LOG(LOG_ERR, "Register a username first\n");
				error_code = JANUS_VIDEOCALL_ERROR_REGISTER_FIRST;
				g_snprintf(error_cause, 512, "Register a username first");
				goto error;
			}
			if(session->peer != NULL) {
				JANUS_LOG(LOG_ERR, "Already in a call\n");
				error_code = JANUS_VIDEOCALL_ERROR_ALREADY_IN_CALL;
				g_snprintf(error_cause, 512, "Already in a call");
				goto error;
			}
			JANUS_VALIDATE_JSON_OBJECT(root, username_parameters,
				error_code, error_cause, TRUE,
				JANUS_VIDEOCALL_ERROR_MISSING_ELEMENT, JANUS_VIDEOCALL_ERROR_INVALID_ELEMENT);
			if(error_code != 0)
				goto error;
			json_t *username = json_object_get(root, "username");
			const char *username_text = json_string_value(username);
			if(!strcmp(username_text, session->username)) {
				JANUS_LOG(LOG_ERR, "You can't call yourself... use the EchoTest for that\n");
				error_code = JANUS_VIDEOCALL_ERROR_USE_ECHO_TEST;
				g_snprintf(error_cause, 512, "You can't call yourself... use the EchoTest for that");
				goto error;
			}
			janus_mutex_lock(&sessions_mutex);
			janus_videocall_session *peer = g_hash_table_lookup(sessions, username_text);
			if(peer == NULL || peer->destroyed) {
				janus_mutex_unlock(&sessions_mutex);
				JANUS_LOG(LOG_ERR, "Username '%s' doesn't exist\n", username_text);
				error_code = JANUS_VIDEOCALL_ERROR_NO_SUCH_USERNAME;
				g_snprintf(error_cause, 512, "Username '%s' doesn't exist", username_text);
				goto error;
			}
			if(peer->peer != NULL) {
				janus_mutex_unlock(&sessions_mutex);
				JANUS_LOG(LOG_VERB, "%s is busy\n", username_text);
				result = json_object();
				json_object_set_new(result, "event", json_string("hangup"));
				json_object_set_new(result, "username", json_string(session->username));
				json_object_set_new(result, "reason", json_string("User busy"));
				/* Also notify event handlers */
				if(notify_events && gateway->events_is_enabled()) {
					json_t *info = json_object();
					json_object_set_new(info, "event", json_string("hangup"));
					json_object_set_new(info, "reason", json_string("User busy"));
					gateway->notify_event(&janus_videocall_plugin, session->handle, info);
				}
				gateway->close_pc(session->handle);
			} else {
				janus_mutex_unlock(&sessions_mutex);
				/* Any SDP to handle? if not, something's wrong */
				if(!msg_sdp) {
					JANUS_LOG(LOG_ERR, "Missing SDP\n");
					error_code = JANUS_VIDEOCALL_ERROR_MISSING_SDP;
					g_snprintf(error_cause, 512, "Missing SDP");
					goto error;
				}
				janus_mutex_lock(&sessions_mutex);
				session->peer = peer;
				peer->peer = session;
				session->has_audio = (strstr(msg_sdp, "m=audio") != NULL);
				session->has_video = (strstr(msg_sdp, "m=video") != NULL);
				session->has_data = (strstr(msg_sdp, "DTLS/SCTP") != NULL);
				janus_mutex_unlock(&sessions_mutex);
				JANUS_LOG(LOG_VERB, "%s is calling %s\n", session->username, session->peer->username);
				JANUS_LOG(LOG_VERB, "This is involving a negotiation (%s) as well:\n%s\n", msg_sdp_type, msg_sdp);
				/* Check if this user will simulcast */
				json_t *msg_simulcast = json_object_get(msg->jsep, "simulcast");
				if(msg_simulcast && janus_get_codec_pt(msg_sdp, "vp8") > 0) {
					JANUS_LOG(LOG_VERB, "VideoCall caller (%s) is going to do simulcasting\n", session->username);
					session->ssrc[0] = json_integer_value(json_object_get(msg_simulcast, "ssrc-0"));
					session->ssrc[1] = json_integer_value(json_object_get(msg_simulcast, "ssrc-1"));
					session->ssrc[2] = json_integer_value(json_object_get(msg_simulcast, "ssrc-2"));
				}
				/* Send SDP to our peer */
				json_t *call = json_object();
				json_object_set_new(call, "videocall", json_string("event"));
				json_t *calling = json_object();
				json_object_set_new(calling, "event", json_string("incomingcall"));
				json_object_set_new(calling, "username", json_string(session->username));
				json_object_set_new(call, "result", calling);
				json_t *jsep = json_pack("{ssss}", "type", msg_sdp_type, "sdp", msg_sdp);
				g_atomic_int_set(&session->hangingup, 0);
				int ret = gateway->push_event(peer->handle, &janus_videocall_plugin, NULL, call, jsep);
				JANUS_LOG(LOG_VERB, "  >> Pushing event to peer: %d (%s)\n", ret, janus_get_api_error(ret));
				json_decref(call);
				json_decref(jsep);
				/* Send an ack back */
				result = json_object();
				json_object_set_new(result, "event", json_string("calling"));
				/* Also notify event handlers */
				if(notify_events && gateway->events_is_enabled()) {
					json_t *info = json_object();
					json_object_set_new(info, "event", json_string("calling"));
					gateway->notify_event(&janus_videocall_plugin, session->handle, info);
				}
			}
		} else if(!strcasecmp(request_text, "accept")) {
			/* Accept a call from another peer */
			if(session->peer == NULL) {
				JANUS_LOG(LOG_ERR, "No incoming call to accept\n");
				error_code = JANUS_VIDEOCALL_ERROR_NO_CALL;
				g_snprintf(error_cause, 512, "No incoming call to accept");
				goto error;
			}
			/* Any SDP to handle? if not, something's wrong */
			if(!msg_sdp) {
				JANUS_LOG(LOG_ERR, "Missing SDP\n");
				error_code = JANUS_VIDEOCALL_ERROR_MISSING_SDP;
				g_snprintf(error_cause, 512, "Missing SDP");
				goto error;
			}
			JANUS_LOG(LOG_VERB, "%s is accepting a call from %s\n", session->username, session->peer->username);
			JANUS_LOG(LOG_VERB, "This is involving a negotiation (%s) as well:\n%s\n", msg_sdp_type, msg_sdp);
			session->has_audio = (strstr(msg_sdp, "m=audio") != NULL);
			session->has_video = (strstr(msg_sdp, "m=video") != NULL);
			session->has_data = (strstr(msg_sdp, "DTLS/SCTP") != NULL);
			/* Check if this user will simulcast */
			json_t *msg_simulcast = json_object_get(msg->jsep, "simulcast");
			if(msg_simulcast && janus_get_codec_pt(msg_sdp, "vp8") > 0) {
				JANUS_LOG(LOG_VERB, "VideoCall callee (%s) is going to do simulcasting\n", session->username);
				session->ssrc[0] = json_integer_value(json_object_get(msg_simulcast, "ssrc-0"));
				session->ssrc[1] = json_integer_value(json_object_get(msg_simulcast, "ssrc-1"));
				session->ssrc[2] = json_integer_value(json_object_get(msg_simulcast, "ssrc-2"));
			} else {
				session->ssrc[0] = 0;
				session->ssrc[1] = 0;
				session->ssrc[2] = 0;
				if(session->peer) {
					session->peer->ssrc[0] = 0;
					session->peer->ssrc[1] = 0;
					session->peer->ssrc[2] = 0;
				}
			}
			/* Send SDP to our peer */
			json_t *jsep = json_pack("{ssss}", "type", msg_sdp_type, "sdp", msg_sdp);
			json_t *call = json_object();
			json_object_set_new(call, "videocall", json_string("event"));
			json_t *calling = json_object();
			json_object_set_…