mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-18 07:47:17 +00:00
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/3892>
This commit is contained in:
parent
04910ac6c1
commit
f6e672f27f
10 changed files with 14 additions and 14 deletions
|
@ -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`
|
||||
|
||||
|
|
|
@ -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)
|
||||
|
|
|
@ -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
|
||||
|
|
|
@ -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,
|
||||
|
|
|
@ -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;
|
||||
|
|
|
@ -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;
|
||||
|
|
|
@ -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)]
|
||||
|
|
|
@ -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)
|
||||
|
|
|
@ -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;
|
||||
|
|
|
@ -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',
|
||||
|
|
Loading…
Reference in a new issue