--- /dev/null
+/*
+ * Demo gstreamer app for negotiating and streaming a sendrecv webrtc stream
+ * with a browser JS app.
+ *
+ * gcc webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o webrtc-sendrecv
+ *
+ * Author: Nirbheek Chauhan <nirbheek@centricular.com>
+ */
+#include <gst/gst.h>
+#include <gst/sdp/sdp.h>
+#include <gst/webrtc/webrtc.h>
+
+/* For signalling */
+#include <libsoup/soup.h>
+#include <json-glib/json-glib.h>
+
+#include <string.h>
+
+enum AppState {
+ APP_STATE_UNKNOWN = 0,
+ APP_STATE_ERROR = 1, /* generic error */
+ SERVER_CONNECTING = 1000,
+ SERVER_CONNECTION_ERROR,
+ SERVER_CONNECTED, /* Ready to register */
+ SERVER_REGISTERING = 2000,
+ SERVER_REGISTRATION_ERROR,
+ SERVER_REGISTERED, /* Ready to call a peer */
+ SERVER_CLOSED, /* server connection closed by us or the server */
+ PEER_CONNECTING = 3000,
+ PEER_CONNECTION_ERROR,
+ PEER_CONNECTED,
+ PEER_CALL_NEGOTIATING = 4000,
+ PEER_CALL_STARTED,
+ PEER_CALL_STOPPING,
+ PEER_CALL_STOPPED,
+ PEER_CALL_ERROR,
+};
+
+static GMainLoop *loop;
+static GstElement *pipe1, *webrtc1;
+
+static SoupWebsocketConnection *ws_conn = NULL;
+static enum AppState app_state = 0;
+static const gchar *peer_id = NULL;
+static const gchar *server_url = "wss://webrtc.nirbheek.in:8443";
+
+static GOptionEntry entries[] =
+{
+ { "peer-id", 0, 0, G_OPTION_ARG_STRING, &peer_id, "String ID of the peer to connect to", "ID" },
+ { "server", 0, 0, G_OPTION_ARG_STRING, &server_url, "Signalling server to connect to", "URL" },
+};
+
+static gboolean
+cleanup_and_quit_loop (const gchar * msg, enum AppState state)
+{
+ if (msg)
+ g_printerr ("%s\n", msg);
+ if (state > 0)
+ app_state = state;
+
+ if (ws_conn) {
+ if (soup_websocket_connection_get_state (ws_conn) ==
+ SOUP_WEBSOCKET_STATE_OPEN)
+ /* This will call us again */
+ soup_websocket_connection_close (ws_conn, 1000, "");
+ else
+ g_object_unref (ws_conn);
+ }
+
+ if (loop) {
+ g_main_loop_quit (loop);
+ loop = NULL;
+ }
+
+ /* To allow usage as a GSourceFunc */
+ return G_SOURCE_REMOVE;
+}
+
+static gchar*
+get_string_from_json_object (JsonObject * object)
+{
+ JsonNode *root;
+ JsonGenerator *generator;
+ gchar *text;
+
+ /* Make it the root node */
+ root = json_node_init_object (json_node_alloc (), object);
+ generator = json_generator_new ();
+ json_generator_set_root (generator, root);
+ text = json_generator_to_data (generator, NULL);
+
+ /* Release everything */
+ g_object_unref (generator);
+ json_node_free (root);
+ return text;
+}
+
+static void
+handle_media_stream (GstPad * pad, GstElement * pipe, const char * convert_name,
+ const char * sink_name)
+{
+ GstPad *qpad;
+ GstElement *q, *conv, *sink;
+ GstPadLinkReturn ret;
+
+ q = gst_element_factory_make ("queue", NULL);
+ g_assert (q);
+ conv = gst_element_factory_make (convert_name, NULL);
+ g_assert (conv);
+ sink = gst_element_factory_make (sink_name, NULL);
+ g_assert (sink);
+ gst_bin_add_many (GST_BIN (pipe), q, conv, sink, NULL);
+ gst_element_sync_state_with_parent (q);
+ gst_element_sync_state_with_parent (conv);
+ gst_element_sync_state_with_parent (sink);
+ gst_element_link_many (q, conv, sink, NULL);
+
+ qpad = gst_element_get_static_pad (q, "sink");
+
+ ret = gst_pad_link (pad, qpad);
+ g_assert (ret == GST_PAD_LINK_OK);
+}
+
+static void
+on_incoming_decodebin_stream (GstElement * decodebin, GstPad * pad,
+ GstElement * pipe)
+{
+ GstCaps *caps;
+ const gchar *name;
+
+ if (!gst_pad_has_current_caps (pad)) {
+ g_printerr ("Pad '%s' has no caps, can't do anything, ignoring\n",
+ GST_PAD_NAME (pad));
+ return;
+ }
+
+ caps = gst_pad_get_current_caps (pad);
+ name = gst_structure_get_name (gst_caps_get_structure (caps, 0));
+
+ if (g_str_has_prefix (name, "video")) {
+ handle_media_stream (pad, pipe, "videoconvert", "autovideosink");
+ } else if (g_str_has_prefix (name, "audio")) {
+ handle_media_stream (pad, pipe, "audioconvert", "autoaudiosink");
+ } else {
+ g_printerr ("Unknown pad %s, ignoring", GST_PAD_NAME (pad));
+ }
+}
+
+static void
+on_incoming_stream (GstElement * webrtc, GstPad * pad, GstElement * pipe)
+{
+ GstElement *decodebin;
+
+ if (GST_PAD_DIRECTION (pad) != GST_PAD_SRC)
+ return;
+
+ decodebin = gst_element_factory_make ("decodebin", NULL);
+ g_signal_connect (decodebin, "pad-added",
+ G_CALLBACK (on_incoming_decodebin_stream), pipe);
+ gst_bin_add (GST_BIN (pipe), decodebin);
+ gst_element_sync_state_with_parent (decodebin);
+ gst_element_link (webrtc, decodebin);
+}
+
+static void
+send_ice_candidate_message (GstElement * webrtc G_GNUC_UNUSED, guint mlineindex,
+ gchar * candidate, gpointer user_data G_GNUC_UNUSED)
+{
+ gchar *text;
+ JsonObject *ice, *msg;
+
+ if (app_state < PEER_CALL_NEGOTIATING) {
+ cleanup_and_quit_loop ("Can't send ICE, not in call", APP_STATE_ERROR);
+ return;
+ }
+
+ ice = json_object_new ();
+ json_object_set_string_member (ice, "candidate", candidate);
+ json_object_set_int_member (ice, "sdpMLineIndex", mlineindex);
+ msg = json_object_new ();
+ json_object_set_object_member (msg, "ice", ice);
+ text = get_string_from_json_object (msg);
+ json_object_unref (msg);
+
+ soup_websocket_connection_send_text (ws_conn, text);
+ g_free (text);
+}
+
+static void
+send_sdp_offer (GstWebRTCSessionDescription * offer)
+{
+ gchar *text;
+ JsonObject *msg, *sdp;
+
+ if (app_state < PEER_CALL_NEGOTIATING) {
+ cleanup_and_quit_loop ("Can't send offer, not in call", APP_STATE_ERROR);
+ return;
+ }
+
+ text = gst_sdp_message_as_text (offer->sdp);
+ g_print ("Sending offer:\n%s\n", text);
+
+ sdp = json_object_new ();
+ json_object_set_string_member (sdp, "type", "offer");
+ json_object_set_string_member (sdp, "sdp", text);
+ g_free (text);
+
+ msg = json_object_new ();
+ json_object_set_object_member (msg, "sdp", sdp);
+ text = get_string_from_json_object (msg);
+ json_object_unref (msg);
+
+ soup_websocket_connection_send_text (ws_conn, text);
+ g_free (text);
+}
+
+/* Offer created by our pipeline, to be sent to the peer */
+static void
+on_offer_received (GstPromise * promise, gpointer user_data)
+{
+ GstWebRTCSessionDescription *offer = NULL;
+ gchar *desc;
+
+ g_assert (app_state == PEER_CALL_NEGOTIATING);
+
+ g_assert (promise->result == GST_PROMISE_RESULT_REPLIED);
+ gst_structure_get (promise->promise, "offer",
+ GST_TYPE_WEBRTC_SESSION_DESCRIPTION, &offer, NULL);
+ gst_promise_unref (promise);
+
+ promise = gst_promise_new ();
+ g_signal_emit_by_name (webrtc1, "set-local-description", offer, promise);
+ gst_promise_interrupt (promise);
+ gst_promise_unref (promise);
+
+ /* Send offer to peer */
+ send_sdp_offer (offer);
+ gst_webrtc_session_description_free (offer);
+}
+
+static void
+on_negotiation_needed (GstElement * element, gpointer user_data)
+{
+ GstPromise *promise = gst_promise_new ();
+
+ app_state = PEER_CALL_NEGOTIATING;
+ g_signal_emit_by_name (webrtc1, "create-offer", NULL, promise);
+ gst_promise_set_change_callback (promise, on_offer_received, user_data,
+ NULL);
+}
+
+#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="
+#define RTP_CAPS_VP8 "application/x-rtp,media=video,encoding-name=VP8,payload="
+
+static gboolean
+start_pipeline (void)
+{
+ GstStateChangeReturn ret;
+ GError *error = NULL;
+
+ pipe1 =
+ gst_parse_launch ("webrtcbin name=sendrecv "
+ "videotestsrc pattern=ball ! queue ! vp8enc deadline=1 ! rtpvp8pay ! "
+ "queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "
+ "audiotestsrc wave=red-noise ! queue ! opusenc ! rtpopuspay ! "
+ "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ",
+ &error);
+
+ if (error) {
+ g_printerr ("Failed to parse launch: %s\n", error->message);
+ g_error_free (error);
+ goto err;
+ }
+
+ webrtc1 = gst_bin_get_by_name (GST_BIN (pipe1), "sendrecv");
+ g_assert (webrtc1 != NULL);
+
+ /* This is the gstwebrtc entry point where we create the offer and so on. It
+ * will be called when the pipeline goes to PLAYING. */
+ g_signal_connect (webrtc1, "on-negotiation-needed",
+ G_CALLBACK (on_negotiation_needed), NULL);
+ /* We need to transmit this ICE candidate to the browser via the websockets
+ * signalling server. Incoming ice candidates from the browser need to be
+ * added by us too, see on_server_message() */
+ g_signal_connect (webrtc1, "on-ice-candidate",
+ G_CALLBACK (send_ice_candidate_message), NULL);
+ /* Incoming streams will be exposed via this signal */
+ g_signal_connect (webrtc1, "pad-added", G_CALLBACK (on_incoming_stream),
+ pipe1);
+ /* Lifetime is the same as the pipeline itself */
+ gst_object_unref (webrtc1);
+
+ g_print ("Starting pipeline\n");
+ ret = gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_PLAYING);
+ if (ret == GST_STATE_CHANGE_FAILURE)
+ goto err;
+
+ return TRUE;
+
+err:
+ if (pipe1)
+ g_clear_object (&pipe1);
+ if (webrtc1)
+ webrtc1 = NULL;
+ return FALSE;
+}
+
+static gboolean
+setup_call (void)
+{
+ gchar *msg;
+
+ if (soup_websocket_connection_get_state (ws_conn) !=
+ SOUP_WEBSOCKET_STATE_OPEN)
+ return FALSE;
+
+ if (!peer_id)
+ return FALSE;
+
+ g_print ("Setting up signalling server call with %s\n", peer_id);
+ app_state = PEER_CONNECTING;
+ msg = g_strdup_printf ("SESSION %s", peer_id);
+ soup_websocket_connection_send_text (ws_conn, msg);
+ g_free (msg);
+ return TRUE;
+}
+
+static gboolean
+register_with_server (void)
+{
+ gchar *hello;
+ gint32 our_id;
+
+ if (soup_websocket_connection_get_state (ws_conn) !=
+ SOUP_WEBSOCKET_STATE_OPEN)
+ return FALSE;
+
+ our_id = g_random_int_range (10, 10000);
+ g_print ("Registering id %i with server\n", our_id);
+ app_state = SERVER_REGISTERING;
+
+ /* Register with the server with a random integer id. Reply will be received
+ * by on_server_message() */
+ hello = g_strdup_printf ("HELLO %i", our_id);
+ soup_websocket_connection_send_text (ws_conn, hello);
+ g_free (hello);
+
+ return TRUE;
+}
+
+static void
+on_server_closed (SoupWebsocketConnection * conn G_GNUC_UNUSED,
+ gpointer user_data G_GNUC_UNUSED)
+{
+ app_state = SERVER_CLOSED;
+ cleanup_and_quit_loop ("Server connection closed", 0);
+}
+
+/* One mega message handler for our asynchronous calling mechanism */
+static void
+on_server_message (SoupWebsocketConnection * conn, SoupWebsocketDataType type,
+ GBytes * message, gpointer user_data)
+{
+ gsize size;
+ gchar *text, *data;
+
+ switch (type) {
+ case SOUP_WEBSOCKET_DATA_BINARY:
+ g_printerr ("Received unknown binary message, ignoring\n");
+ g_bytes_unref (message);
+ return;
+ case SOUP_WEBSOCKET_DATA_TEXT:
+ data = g_bytes_unref_to_data (message, &size);
+ /* Convert to NULL-terminated string */
+ text = g_strndup (data, size);
+ g_free (data);
+ break;
+ default:
+ g_assert_not_reached ();
+ }
+
+ /* Server has accepted our registration, we are ready to send commands */
+ if (g_strcmp0 (text, "HELLO") == 0) {
+ if (app_state != SERVER_REGISTERING) {
+ cleanup_and_quit_loop ("ERROR: Received HELLO when not registering",
+ APP_STATE_ERROR);
+ goto out;
+ }
+ app_state = SERVER_REGISTERED;
+ g_print ("Registered with server\n");
+ /* Ask signalling server to connect us with a specific peer */
+ if (!setup_call ()) {
+ cleanup_and_quit_loop ("ERROR: Failed to setup call", PEER_CALL_ERROR);
+ goto out;
+ }
+ /* Call has been setup by the server, now we can start negotiation */
+ } else if (g_strcmp0 (text, "SESSION_OK") == 0) {
+ if (app_state != PEER_CONNECTING) {
+ cleanup_and_quit_loop ("ERROR: Received SESSION_OK when not calling",
+ PEER_CONNECTION_ERROR);
+ goto out;
+ }
+
+ app_state = PEER_CONNECTED;
+ /* Start negotiation (exchange SDP and ICE candidates) */
+ if (!start_pipeline ())
+ cleanup_and_quit_loop ("ERROR: failed to start pipeline",
+ PEER_CALL_ERROR);
+ /* Handle errors */
+ } else if (g_str_has_prefix (text, "ERROR")) {
+ switch (app_state) {
+ case SERVER_CONNECTING:
+ app_state = SERVER_CONNECTION_ERROR;
+ break;
+ case SERVER_REGISTERING:
+ app_state = SERVER_REGISTRATION_ERROR;
+ break;
+ case PEER_CONNECTING:
+ app_state = PEER_CONNECTION_ERROR;
+ break;
+ case PEER_CONNECTED:
+ case PEER_CALL_NEGOTIATING:
+ app_state = PEER_CALL_ERROR;
+ default:
+ app_state = APP_STATE_ERROR;
+ }
+ cleanup_and_quit_loop (text, 0);
+ /* Look for JSON messages containing SDP and ICE candidates */
+ } else {
+ JsonNode *root;
+ JsonObject *object;
+ JsonParser *parser = json_parser_new ();
+ if (!json_parser_load_from_data (parser, text, -1, NULL)) {
+ g_printerr ("Unknown message '%s', ignoring", text);
+ g_object_unref (parser);
+ goto out;
+ }
+
+ root = json_parser_get_root (parser);
+ if (!JSON_NODE_HOLDS_OBJECT (root)) {
+ g_printerr ("Unknown json message '%s', ignoring", text);
+ g_object_unref (parser);
+ goto out;
+ }
+
+ object = json_node_get_object (root);
+ /* Check type of JSON message */
+ if (json_object_has_member (object, "sdp")) {
+ int ret;
+ const gchar *text;
+ GstSDPMessage *sdp;
+ GstWebRTCSessionDescription *answer;
+
+ g_assert (app_state == PEER_CALL_NEGOTIATING);
+
+ g_assert (json_object_has_member (object, "type"));
+ /* In this example, we always create the offer and receive one answer.
+ * See tests/examples/webrtcbidirectional.c in gst-plugins-bad for how to
+ * handle offers from peers and reply with answers using webrtcbin. */
+ g_assert_cmpstr (json_object_get_string_member (object, "type"), ==,
+ "answer");
+
+ text = json_object_get_string_member (object, "sdp");
+
+ g_print ("Received answer:\n%s\n", text);
+
+ ret = gst_sdp_message_new (&sdp);
+ g_assert (ret == GST_SDP_OK);
+
+ ret = gst_sdp_message_parse_buffer (text, strlen (text), sdp);
+ g_assert (ret == GST_SDP_OK);
+
+ answer = gst_webrtc_session_description_new (GST_WEBRTC_SDP_TYPE_ANSWER,
+ sdp);
+ g_assert (answer);
+
+ /* Set remote description on our pipeline */
+ {
+ GstPromise *promise = gst_promise_new ();
+ g_signal_emit_by_name (webrtc1, "set-remote-description", answer,
+ promise);
+ gst_promise_interrupt (promise);
+ gst_promise_unref (promise);
+ }
+
+ app_state = PEER_CALL_STARTED;
+ } else if (json_object_has_member (object, "ice")) {
+ JsonObject *ice;
+ const gchar *candidate;
+ gint sdpmlineindex;
+
+ ice = json_object_get_object_member (object, "ice");
+ candidate = json_object_get_string_member (ice, "candidate");
+ sdpmlineindex = json_object_get_int_member (ice, "sdpMLineIndex");
+
+ /* Add ice candidate sent by remote peer */
+ g_signal_emit_by_name (webrtc1, "add-ice-candidate", sdpmlineindex,
+ candidate);
+ } else {
+ g_printerr ("Ignoring unknown JSON message:\n%s\n", text);
+ }
+ g_object_unref (parser);
+ }
+
+out:
+ g_free (text);
+}
+
+static void
+on_server_connected (SoupSession * session, GAsyncResult * res,
+ SoupMessage *msg)
+{
+ GError *error = NULL;
+
+ ws_conn = soup_session_websocket_connect_finish (session, res, &error);
+ if (error) {
+ cleanup_and_quit_loop (error->message, SERVER_CONNECTION_ERROR);
+ g_error_free (error);
+ return;
+ }
+
+ g_assert (ws_conn != NULL);
+
+ app_state = SERVER_CONNECTED;
+ g_print ("Connected to signalling server\n");
+
+ g_signal_connect (ws_conn, "closed", G_CALLBACK (on_server_closed), NULL);
+ g_signal_connect (ws_conn, "message", G_CALLBACK (on_server_message), NULL);
+
+ /* Register with the server so it knows about us and can accept commands */
+ register_with_server ();
+}
+
+/*
+ * Connect to the signalling server. This is the entrypoint for everything else.
+ */
+static void
+connect_to_websocket_server_async (void)
+{
+ SoupLogger *logger;
+ SoupMessage *message;
+ SoupSession *session;
+ const char *https_aliases[] = {"wss", NULL};
+
+ session = soup_session_new_with_options (SOUP_SESSION_SSL_STRICT, TRUE,
+ SOUP_SESSION_SSL_USE_SYSTEM_CA_FILE, TRUE,
+ //SOUP_SESSION_SSL_CA_FILE, "/etc/ssl/certs/ca-bundle.crt",
+ SOUP_SESSION_HTTPS_ALIASES, https_aliases, NULL);
+
+ logger = soup_logger_new (SOUP_LOGGER_LOG_BODY, -1);
+ soup_session_add_feature (session, SOUP_SESSION_FEATURE (logger));
+ g_object_unref (logger);
+
+ message = soup_message_new (SOUP_METHOD_GET, server_url);
+
+ g_print ("Connecting to server...\n");
+
+ /* Once connected, we will register */
+ soup_session_websocket_connect_async (session, message, NULL, NULL, NULL,
+ (GAsyncReadyCallback) on_server_connected, message);
+ app_state = SERVER_CONNECTING;
+}
+
+int
+main (int argc, char *argv[])
+{
+ SoupSession *session;
+ GOptionContext *context;
+ GError *error = NULL;
+
+ context = g_option_context_new ("- gstreamer webrtc sendrecv demo");
+ g_option_context_add_main_entries (context, entries, NULL);
+ g_option_context_add_group (context, gst_init_get_option_group ());
+ if (!g_option_context_parse (context, &argc, &argv, &error)) {
+ g_printerr ("Error initializing: %s\n", error->message);
+ return -1;
+ }
+
+ if (!peer_id) {
+ g_printerr ("--peer-id is a required argument\n");
+ return -1;
+ }
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+ connect_to_websocket_server_async ();
+
+ g_main_loop_run (loop);
+
+ gst_element_set_state (GST_ELEMENT (pipe1), GST_STATE_NULL);
+ g_print ("Pipeline stopped\n");
+
+ gst_object_unref (pipe1);
+
+ return 0;
+}
--- /dev/null
+/* vim: set sts=4 sw=4 et :
+ *
+ * Demo Javascript app for negotiating and streaming a sendrecv webrtc stream
+ * with a GStreamer app. Runs only in passive mode, i.e., responds to offers
+ * with answers, exchanges ICE candidates, and streams.
+ *
+ * Author: Nirbheek Chauhan <nirbheek@centricular.com>
+ */
+
+var connect_attempts = 0;
+
+var peer_connection = null;
+var rtc_configuration = {iceServers: [{urls: "stun:stun.services.mozilla.com"},
+ {urls: "stun:stun.l.google.com:19302"}]};
+var ws_conn;
+var local_stream;
+var peer_id;
+
+function getOurId() {
+ return Math.floor(Math.random() * (9000 - 10) + 10).toString();
+}
+
+function resetState() {
+ // This will call onServerClose()
+ ws_conn.close();
+}
+
+function handleIncomingError(error) {
+ setStatus("ERROR: " + error);
+ resetState();
+}
+
+function getVideoElement() {
+ return document.getElementById("stream");
+}
+
+function setStatus(text) {
+ console.log(text);
+ document.getElementById("status").textContent = text;
+}
+
+function resetVideoElement() {
+ var videoElement = getVideoElement();
+ videoElement.pause();
+ videoElement.src = "";
+ videoElement.load();
+}
+
+// SDP offer received from peer, set remote description and create an answer
+function onIncomingSDP(sdp) {
+ console.log("Incoming SDP is "+ JSON.stringify(sdp));
+ peer_connection.setRemoteDescription(sdp).then(() => {
+ setStatus("Remote SDP set");
+ if (sdp.type != "offer")
+ return;
+ setStatus("Got SDP offer, creating answer");
+ peer_connection.createAnswer().then(onLocalDescription).catch(setStatus);
+ }).catch(setStatus);
+}
+
+// Local description was set, send it to peer
+function onLocalDescription(desc) {
+ console.log("Got local description: " + JSON.stringify(desc));
+ peer_connection.setLocalDescription(desc).then(function() {
+ setStatus("Sending SDP answer");
+ ws_conn.send(JSON.stringify(peer_connection.localDescription));
+ });
+}
+
+// ICE candidate received from peer, add it to the peer connection
+function onIncomingICE(ice) {
+ console.log("Incoming ICE: " + JSON.stringify(ice));
+ var candidate = new RTCIceCandidate(ice);
+ peer_connection.addIceCandidate(candidate).catch(setStatus);
+}
+
+function onServerMessage(event) {
+ console.log("Received " + event.data);
+ switch (event.data) {
+ case "HELLO":
+ setStatus("Registered with server, waiting for call");
+ return;
+ default:
+ if (event.data.startsWith("ERROR")) {
+ handleIncomingError(event.data);
+ return;
+ }
+ // Handle incoming JSON SDP and ICE messages
+ try {
+ msg = JSON.parse(event.data);
+ } catch (e) {
+ if (e instanceof SyntaxError) {
+ handleIncomingError("Error parsing incoming JSON: " + event.data);
+ } else {
+ handleIncomingError("Unknown error parsing response: " + event.data);
+ }
+ return;
+ }
+
+ // Incoming JSON signals the beginning of a call
+ if (peer_connection == null)
+ createCall(msg);
+
+ if (msg.sdp != null) {
+ onIncomingSDP(msg.sdp);
+ } else if (msg.ice != null) {
+ onIncomingICE(msg.ice);
+ } else {
+ handleIncomingError("Unknown incoming JSON: " + msg);
+ }
+ }
+}
+
+function onServerClose(event) {
+ resetVideoElement();
+
+ if (peer_connection != null) {
+ peer_connection.close();
+ peer_connection = null;
+ }
+
+ // Reset after a second
+ window.setTimeout(websocketServerConnect, 1000);
+}
+
+function onServerError(event) {
+ setStatus("Unable to connect to server, did you add an exception for the certificate?")
+ // Retry after 3 seconds
+ window.setTimeout(websocketServerConnect, 3000);
+}
+
+function websocketServerConnect() {
+ connect_attempts++;
+ if (connect_attempts > 3) {
+ setStatus("Too many connection attempts, aborting. Refresh page to try again");
+ return;
+ }
+ peer_id = getOurId();
+ setStatus("Connecting to server");
+ ws_conn = new WebSocket('wss://' + window.location.hostname + ':8443');
+ /* When connected, immediately register with the server */
+ ws_conn.addEventListener('open', (event) => {
+ document.getElementById("peer-id").textContent = peer_id;
+ ws_conn.send('HELLO ' + peer_id);
+ setStatus("Registering with server");
+ });
+ ws_conn.addEventListener('error', onServerError);
+ ws_conn.addEventListener('message', onServerMessage);
+ ws_conn.addEventListener('close', onServerClose);
+
+ var constraints = {video: true, audio: true};
+
+ // Add local stream
+ if (navigator.mediaDevices.getUserMedia) {
+ navigator.mediaDevices.getUserMedia(constraints)
+ .then((stream) => { local_stream = stream })
+ .catch(errorUserMediaHandler);
+ } else {
+ errorUserMediaHandler();
+ }
+}
+
+function onRemoteStreamAdded(event) {
+ videoTracks = event.stream.getVideoTracks();
+ audioTracks = event.stream.getAudioTracks();
+
+ if (videoTracks.length > 0) {
+ console.log('Incoming stream: ' + videoTracks.length + ' video tracks and ' + audioTracks.length + ' audio tracks');
+ getVideoElement().srcObject = event.stream;
+ } else {
+ handleIncomingError('Stream with unknown tracks added, resetting');
+ }
+}
+
+function errorUserMediaHandler() {
+ setStatus("Browser doesn't support getUserMedia!");
+}
+
+function createCall(msg) {
+ // Reset connection attempts because we connected successfully
+ connect_attempts = 0;
+
+ peer_connection = new RTCPeerConnection(rtc_configuration);
+ peer_connection.onaddstream = onRemoteStreamAdded;
+ /* Send our video/audio to the other peer */
+ peer_connection.addStream(local_stream);
+
+ if (!msg.sdp) {
+ console.log("WARNING: First message wasn't an SDP message!?");
+ }
+
+ peer_connection.onicecandidate = (event) => {
+ // We have a candidate, send it to the remote party with the
+ // same uuid
+ if (event.candidate == null) {
+ console.log("ICE Candidate was null, done");
+ return;
+ }
+ ws_conn.send(JSON.stringify({'ice': event.candidate}));
+ };
+
+ setStatus("Created peer connection for call, waiting for SDP");
+}
projects / platform / upstream / gstreamer.git / commitdiff