diff --git a/webrtc/sendrecv/gst/webrtc-sendrecv.c b/webrtc/sendrecv/gst/webrtc-sendrecv.c
index 109b14d248..321f8d0720 100644
--- a/webrtc/sendrecv/gst/webrtc-sendrecv.c
+++ b/webrtc/sendrecv/gst/webrtc-sendrecv.c
@@ -40,6 +40,7 @@ enum AppState {
static GMainLoop *loop;
static GstElement *pipe1, *webrtc1;
+static GObject *send_channel, *receive_channel;
static SoupWebsocketConnection *ws_conn = NULL;
static enum AppState app_state = 0;
@@ -274,6 +275,54 @@ on_negotiation_needed (GstElement * element, gpointer user_data)
#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 void
+data_channel_on_error (GObject * dc, gpointer user_data)
+{
+ cleanup_and_quit_loop ("Data channel error", 0);
+}
+
+static void
+data_channel_on_open (GObject * dc, gpointer user_data)
+{
+ GBytes *bytes = g_bytes_new ("data", strlen("data"));
+ g_print ("data channel opened\n");
+ g_signal_emit_by_name (dc, "send-string", "Hi! from GStreamer");
+ g_signal_emit_by_name (dc, "send-data", bytes);
+ g_bytes_unref (bytes);
+}
+
+static void
+data_channel_on_close (GObject * dc, gpointer user_data)
+{
+ cleanup_and_quit_loop ("Data channel closed", 0);
+}
+
+static void
+data_channel_on_message_string (GObject * dc, gchar *str, gpointer user_data)
+{
+ g_print ("Received data channel message: %s\n", str);
+}
+
+static void
+connect_data_channel_signals (GObject * data_channel)
+{
+ g_signal_connect (data_channel, "on-error", G_CALLBACK (data_channel_on_error),
+ NULL);
+ g_signal_connect (data_channel, "on-open", G_CALLBACK (data_channel_on_open),
+ NULL);
+ g_signal_connect (data_channel, "on-close", G_CALLBACK (data_channel_on_close),
+ NULL);
+ g_signal_connect (data_channel, "on-message-string", G_CALLBACK (data_channel_on_message_string),
+ NULL);
+}
+
+static void
+on_data_channel (GstElement * webrtc, GObject * data_channel, gpointer user_data)
+{
+ connect_data_channel_signals (data_channel);
+ receive_channel = data_channel;
+}
+
static gboolean
start_pipeline (void)
{
@@ -306,6 +355,17 @@ start_pipeline (void)
* added by us too, see on_server_message() */
g_signal_connect (webrtc1, "on-ice-candidate",
G_CALLBACK (send_ice_candidate_message), NULL);
+ g_signal_emit_by_name (webrtc1, "create-data-channel", "channel", NULL,
+ &send_channel);
+ if (send_channel) {
+ g_print ("Created data channel\n");
+ connect_data_channel_signals (send_channel);
+ } else {
+ g_print ("Could not create data channel, is usrsctp available?\n");
+ }
+
+ g_signal_connect (webrtc1, "on-data-channel", G_CALLBACK (on_data_channel),
+ NULL);
/* Incoming streams will be exposed via this signal */
g_signal_connect (webrtc1, "pad-added", G_CALLBACK (on_incoming_stream),
pipe1);
diff --git a/webrtc/sendrecv/js/index.html b/webrtc/sendrecv/js/index.html
index f52cd507ec..9a95fb7491 100644
--- a/webrtc/sendrecv/js/index.html
+++ b/webrtc/sendrecv/js/index.html
@@ -24,6 +24,7 @@
Status: unknown
+
Our id is unknown
diff --git a/webrtc/sendrecv/js/webrtc.js b/webrtc/sendrecv/js/webrtc.js
index b94288f825..c13837e4c5 100644
--- a/webrtc/sendrecv/js/webrtc.js
+++ b/webrtc/sendrecv/js/webrtc.js
@@ -20,6 +20,7 @@ var default_constraints = {video: true, audio: true};
var connect_attempts = 0;
var peer_connection;
+var send_channel;
var ws_conn;
// Promise for local stream after constraints are approved by the user
var local_stream_promise;
@@ -234,6 +235,41 @@ function errorUserMediaHandler() {
setError("Browser doesn't support getUserMedia!");
}
+const handleDataChannelOpen = (event) =>{
+ console.log("dataChannel.OnOpen", event);
+};
+
+const handleDataChannelMessageReceived = (event) =>{
+ console.log("dataChannel.OnMessage:", event, event.data.type);
+
+ setStatus("Received data channel message");
+ if (typeof event.data === 'string' || event.data instanceof String) {
+ console.log('Incoming string message: ' + event.data);
+ textarea = document.getElementById("text")
+ textarea.value = textarea.value + '\n' + event.data
+ } else {
+ console.log('Incoming data message');
+ }
+ send_channel.send("Hi! (from browser)");
+};
+
+const handleDataChannelError = (error) =>{
+ console.log("dataChannel.OnError:", error);
+};
+
+const handleDataChannelClose = (event) =>{
+ console.log("dataChannel.OnClose", event);
+};
+
+function onDataChannel(event) {
+ setStatus("Data channel created");
+ let receiveChannel = event.channel;
+ receiveChannel.onopen = handleDataChannelOpen;
+ receiveChannel.onmessage = handleDataChannelMessageReceived;
+ receiveChannel.onerror = handleDataChannelError;
+ receiveChannel.onclose = handleDataChannelClose;
+}
+
function createCall(msg) {
// Reset connection attempts because we connected successfully
connect_attempts = 0;
@@ -241,6 +277,12 @@ function createCall(msg) {
console.log('Creating RTCPeerConnection');
peer_connection = new RTCPeerConnection(rtc_configuration);
+ send_channel = peer_connection.createDataChannel('label', null);
+ send_channel.onopen = handleDataChannelOpen;
+ send_channel.onmessage = handleDataChannelMessageReceived;
+ send_channel.onerror = handleDataChannelError;
+ send_channel.onclose = handleDataChannelClose;
+ peer_connection.ondatachannel = onDataChannel;
peer_connection.onaddstream = onRemoteStreamAdded;
/* Send our video/audio to the other peer */
local_stream_promise = getLocalStream().then((stream) => {