webrtc examples: Use webrtc.gstreamer.net

[platform/upstream/gstreamer.git] / subprojects / gst-examples / webrtc / sendrecv / gst / webrtc-sendrecv.c

diff --git a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c
index 9455e26..6263b0e 100644 (file)
--- a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c
+++ b/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;
@@ -481,7 +481,8 @@ start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
   audio_desc =
       g_strdup_printf
       ("audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample"
-      "! queue ! opusenc ! rtpopuspay name=audiopay pt=%u ! queue", opus_pt);
+      "! queue ! opusenc ! rtpopuspay name=audiopay pt=%u "
+      "! application/x-rtp, encoding-name=OPUS ! queue", opus_pt);
   audio_bin = gst_parse_bin_from_description (audio_desc, TRUE, &audio_error);
   g_free (audio_desc);
   if (audio_error) {