X-Git-Url: https://git.toastfreeware.priv.at/toast/stream2beamer.git/blobdiff_plain/c7973a1e48b8ff169d4d6e5585df55431de43cc3..7fc8b5a1f9425a11ccf996b578ca5f907722d786:/laplace_client.py?ds=inline

diff --git a/laplace_client.py b/laplace_client.py
index 2778341..1b9c8f0 100755
--- a/laplace_client.py
+++ b/laplace_client.py
@@ -31,6 +31,7 @@ class WebRTCClient:
         self.ssl_context.verify_mode = ssl.CERT_NONE
         self.websocket = None
         self.session_id = None
+        self.userfragments = []
 
     def send_sdp_offer(self, offer):
         text = offer.sdp.as_text()
@@ -57,6 +58,9 @@ class WebRTCClient:
         promise.interrupt()
         self.send_sdp_offer(offer)
 
+        sdp = offer.sdp
+        self.userfragments = [sdp.get_media(i).get_attribute_val('ice-ufrag') for i in range(sdp.medias_len())]
+
     def on_negotiation_needed(self, element):
         log.info('on_negotiation_needed')
         promise = Gst.Promise.new_with_change_func(self.on_offer_created, element, None)
@@ -68,8 +72,9 @@ class WebRTCClient:
             'Type': 'addCalleeIceCandidate',
             'Value': json.dumps({
                 "candidate": candidate,
-                "sdpMid": "0",
+                "sdpMid": f"{mlineindex}",
                 "sdpMLineIndex": mlineindex,
+                "usernameFragment": self.userfragments[mlineindex],
                 })
             })
         log.info(f'send_ice_candidate_message with {icemsg}')
@@ -120,12 +125,12 @@ class WebRTCClient:
 
     def start_pipeline(self):
         self.webrtc = Gst.ElementFactory.make('webrtcbin', 'laplace')
+        # self.webrtc.set_property("bundle-policy", 3)
         direction = GstWebRTC.WebRTCRTPTransceiverDirection.RECVONLY
-        vcaps = Gst.caps_from_string("application/x-rtp,media=video,encoding-name=VP8/9000,payload=96")
-        acaps = Gst.caps_from_string("application/x-rtp,media=audio,encoding-name=OPUS,media=audio,clock-rate=48000,payload=97")
-        # vcaps.append(acaps)
-        self.webrtc.emit('add-transceiver', direction, vcaps)
-        self.webrtc.emit('add-transceiver', direction, acaps)
+        video_caps = Gst.caps_from_string("application/x-rtp,media=video,encoding-name=VP8/9000,payload=96")
+        audio_caps = Gst.caps_from_string("application/x-rtp,media=audio,encoding-name=OPUS,clock-rate=48000,payload=111")
+        self.webrtc.emit('add-transceiver', direction, video_caps)
+        self.webrtc.emit('add-transceiver', direction, audio_caps)
         self.pipe = Gst.Pipeline.new("pipeline")
         Gst.Bin.do_add_element(self.pipe, self.webrtc)
         self.webrtc.connect('on-negotiation-needed', self.on_negotiation_needed)
@@ -193,7 +198,7 @@ def main():
     if not check_plugins():
         sys.exit(1)
     parser = argparse.ArgumentParser()
-    parser.add_argument('--uri', default='wss://localhost:2222/ws_connect?id=cug',
+    parser.add_argument('--uri', default='wss://localhost:1234/ws_connect?id=cug',
         help='Signalling server URI')
     args = parser.parse_args()
     c = WebRTCClient(args.uri)