webrtc examples: Use webrtc.gstreamer.net

Actually just a CNAME to webrtc.nirbheek.in for now, but it allows
replacement / hosting without my involvement, so reduces the bus
factor.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3802>

subprojects/gst-examples/webrtc/README.md

@@ -61,7 +61,7 @@ Build outputs will be placed in the directory `_builddir`.
 
 ### sendrecv: Send and receive audio and video
 
-* Serve the `js/` directory on the root of your website, or open https://webrtc.nirbheek.in
+* Serve the `js/` directory on the root of your website, or open https://webrtc.gstreamer.net
   - The JS code assumes the signalling server is on port 8443 of the same server serving the HTML
 
 * Open the website in a browser and ensure that the status is "Registered with server, waiting for call", and note the `id` too.
@@ -93,7 +93,7 @@ You can pass a --server argument to all versions, for example `--server=wss://12
 `./gradlew build`\
 `java -jar build/libs/gst-java.jar --peer-id=ID` with the `id` from the browser.
 
-You can optionally specify the server URL too (it defaults to wss://webrtc.nirbheek.in:8443):
+You can optionally specify the server URL too (it defaults to wss://webrtc.gstreamer.net:8443):
 
 `java -jar build/libs/gst-java.jar --peer-id=1 --server=ws://localhost:8443`
 

subprojects/gst-examples/webrtc/android/app/src/main/jni/webrtc.c

@@ -38,7 +38,7 @@
 GST_DEBUG_CATEGORY_STATIC (debug_category);
 #define GST_CAT_DEFAULT debug_category
 
-#define DEFAULT_SIGNALLING_SERVER "wss://webrtc.nirbheek.in:8443"
+#define DEFAULT_SIGNALLING_SERVER "wss://webrtc.gstreamer.net:8443"
 
 #define GET_CUSTOM_DATA(env, thiz, fieldID) (WebRTC *)(gintptr)(*env)->GetLongField (env, thiz, fieldID)
 #define SET_CUSTOM_DATA(env, thiz, fieldID, data) (*env)->SetLongField (env, thiz, fieldID, (jlong)(gintptr)data)

subprojects/gst-examples/webrtc/android/app/src/main/res/layout/main.xml

@@ -38,7 +38,7 @@
                 android:layout_height="wrap_content"
                 android:layout_margin="8dp"
                 android:inputType="textUri"
-                android:text="wss://webrtc.nirbheek.in:8443" />
+                android:text="wss://webrtc.gstreamer.net:8443" />
         </TableRow>
 
         <TableRow

subprojects/gst-examples/webrtc/multiparty-sendrecv/gst-rust/src/main.rs

@@ -25,7 +25,7 @@ use serde_derive::{Deserialize, Serialize};
 use anyhow::{anyhow, bail, Context};
 
 const STUN_SERVER: &str = "stun://stun.l.google.com:19302";
-const TURN_SERVER: &str = "turn://foo:bar@webrtc.nirbheek.in:3478";
+const TURN_SERVER: &str = "turn://foo:bar@webrtc.gstreamer.net:3478";
 const VIDEO_WIDTH: u32 = 1024;
 const VIDEO_HEIGHT: u32 = 768;
 
@@ -45,7 +45,7 @@ macro_rules! upgrade_weak {
 
 #[derive(Debug, clap::Parser)]
 struct Args {
-    #[clap(short, long, default_value = "wss://webrtc.nirbheek.in:8443")]
+    #[clap(short, long, default_value = "wss://webrtc.gstreamer.net:8443")]
     server: String,
     #[clap(short, long)]
     room_id: u32,

subprojects/gst-examples/webrtc/multiparty-sendrecv/gst/mp-webrtc-sendrecv.c

@@ -46,7 +46,7 @@ static GList *peers;
 
 static SoupWebsocketConnection *ws_conn = NULL;
 static enum AppState app_state = 0;
-static const gchar *default_server_url = "wss://webrtc.nirbheek.in:8443";
+static const gchar *default_server_url = "wss://webrtc.gstreamer.net:8443";
 static gchar *server_url = NULL;
 static gchar *local_id = NULL;
 static gchar *room_id = NULL;

subprojects/gst-examples/webrtc/sendrecv/gst-java/src/main/java/WebrtcSendRecv.java

@@ -28,7 +28,7 @@ import java.io.IOException;
 public class WebrtcSendRecv {
 
     private static final Logger logger = LoggerFactory.getLogger(WebrtcSendRecv.class);
-    private static final String REMOTE_SERVER_URL = "wss://webrtc.nirbheek.in:8443";
+    private static final String REMOTE_SERVER_URL = "wss://webrtc.gstreamer.net:8443";
     private static final String VIDEO_BIN_DESCRIPTION = "videotestsrc ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay ! queue ! capsfilter caps=application/x-rtp,media=video,encoding-name=VP8,payload=97";
     private static final String AUDIO_BIN_DESCRIPTION = "audiotestsrc ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! queue ! capsfilter caps=application/x-rtp,media=audio,encoding-name=OPUS,payload=96";
 
@@ -41,7 +41,7 @@ public class WebrtcSendRecv {
 
     public static void main(String[] args) throws Exception {
         if (args.length == 0) {
-            logger.error("Please pass at least the peer-id from the signalling server e.g java -jar build/libs/gst-java.jar --peer-id=1234 --server=wss://webrtc.nirbheek.in:8443");
+            logger.error("Please pass at least the peer-id from the signalling server e.g java -jar build/libs/gst-java.jar --peer-id=1234 --server=wss://webrtc.gstreamer.net:8443");
             return;
         }
         String serverUrl = REMOTE_SERVER_URL;

subprojects/gst-examples/webrtc/sendrecv/gst-rust/src/main.rs

@@ -25,7 +25,7 @@ use serde_derive::{Deserialize, Serialize};
 use anyhow::{anyhow, bail, Context};
 
 const STUN_SERVER: &str = "stun://stun.l.google.com:19302";
-const TURN_SERVER: &str = "turn://foo:bar@webrtc.nirbheek.in:3478";
+const TURN_SERVER: &str = "turn://foo:bar@webrtc.gstreamer.net:3478";
 
 const TWCC_URI: &str = "http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01";
 
@@ -45,7 +45,7 @@ macro_rules! upgrade_weak {
 
 #[derive(Debug, clap::Parser)]
 struct Args {
-    #[clap(short, long, default_value = "wss://webrtc.nirbheek.in:8443")]
+    #[clap(short, long, default_value = "wss://webrtc.gstreamer.net:8443")]
     server: String,
     /// Peer ID that should be called. If not given then an incoming call is expected.
     #[clap(short, long)]

subprojects/gst-examples/webrtc/sendrecv/gst-sharp/WebRTCSendRecv.cs

@@ -284,7 +284,7 @@ namespace GstWebRTCDemo
 
     static class WebRtcSendRcv
     {
-        const string SERVER = "wss://webrtc.nirbheek.in:8443";
+        const string SERVER = "wss://webrtc.gstreamer.net:8443";
         static Random random = new Random();
 
         public static void Main(string[] args)

subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c

@@ -53,7 +53,7 @@ static SoupWebsocketConnection *ws_conn = NULL;
 static enum AppState app_state = 0;
 static gchar *peer_id = NULL;
 static gchar *our_id = NULL;
-static const gchar *server_url = "wss://webrtc.nirbheek.in:8443";
+static const gchar *server_url = "wss://webrtc.gstreamer.net:8443";
 static gboolean disable_ssl = FALSE;
 static gboolean remote_is_offerer = FALSE;
 static gboolean custom_ice = FALSE;

subprojects/gst-examples/webrtc/sendrecv/gst/webrtc_sendrecv.py

@@ -353,7 +353,7 @@ if __name__ == '__main__':
                         help='Video encoding to negotiate')
     parser.add_argument('--peer-id', help='String ID of the peer to connect to')
     parser.add_argument('--our-id', help='String ID that the peer can use to connect to us')
-    parser.add_argument('--server', default='wss://webrtc.nirbheek.in:8443',
+    parser.add_argument('--server', default='wss://webrtc.gstreamer.net:8443',
                         help='Signalling server to connect to, eg "wss://127.0.0.1:8443"')
     parser.add_argument('--remote-offerer', default=False, action='store_true',
                         dest='remote_is_offerer',