Add support for creating the offer in the Rust sendrecv client

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

@@ -81,7 +81,8 @@ struct AppWeak(Weak<AppInner>);
 // Actual application state
 #[derive(Debug)]
 struct AppInner {
-    peer_id: String,
+    // None if we wait for a peer to appear
+    peer_id: Option<String>,
     pipeline: gst::Pipeline,
     webrtcbin: gst::Element,
     send_msg_tx: Mutex<mpsc::UnboundedSender<OwnedMessage>>,
@@ -202,6 +203,46 @@ impl App {
         self.send_sdp_offer(&offer)
     }
 
+    // Send our SDP answer via the WebSocket connection to the peer as JSON message
+    fn send_sdp_answer(&self, offer: &gst_webrtc::WebRTCSessionDescription) -> Result<(), Error> {
+        let message = serde_json::to_string(&JsonMsg::Sdp {
+            type_: "answer".to_string(),
+            sdp: offer.get_sdp().as_text().unwrap(),
+        })
+        .unwrap();
+
+        println!("Sending SDP answer to peer: {}", message);
+
+        self.send_text_msg(message)
+    }
+
+    // Once webrtcbin has create the answer SDP for us, handle it by sending it to the peer via the
+    // WebSocket connection
+    fn on_answer_created(&self, promise: &gst::Promise) -> Result<(), Error> {
+        let reply = match promise.wait() {
+            gst::PromiseResult::Replied => promise.get_reply().unwrap(),
+            err => {
+                return Err(GStreamerError(format!(
+                    "Offer creation future got no reponse: {:?}",
+                    err
+                ))
+                .into());
+            }
+        };
+
+        let answer = reply
+            .get_value("answer")
+            .unwrap()
+            .get::<gst_webrtc::WebRTCSessionDescription>()
+            .expect("Invalid argument");
+        self.0
+            .webrtcbin
+            .emit("set-local-description", &[&answer, &None::<gst::Promise>])
+            .unwrap();
+
+        self.send_sdp_answer(&answer)
+    }
+
     // Whenever webrtcbin tells us that (re-)negotiation is needed, simply ask
     // for a new offer SDP from webrtcbin without any customization and then
     // asynchronously send it to the peer via the WebSocket connection
@@ -405,14 +446,16 @@ impl App {
 
     // Finish creating our pipeline and actually start it once the connection with the peer is
     // there
-    fn start_pipeline(&self) -> Result<(), Error> {
+    fn setup_pipeline(&self) -> Result<(), Error> {
         println!("Start pipeline");
 
         // Create our audio/video sources we send to the peer
         self.add_video_source()?;
         self.add_audio_source()?;
 
-        // Whenever (re-)negotiation is needed, do so
+        // Whenever (re-)negotiation is needed, do so but this is only needed if
+        // we send the initial offer
+        if self.0.peer_id.is_some() {
             let app_clone = self.downgrade();
             self.0
                 .webrtcbin
@@ -428,6 +471,7 @@ impl App {
                     None
                 })
                 .unwrap();
+        }
 
         // Whenever there is a new ICE candidate, send it to the peer
         let app_clone = self.downgrade();
@@ -458,6 +502,29 @@ impl App {
             }
         });
 
+        Ok(())
+    }
+
+    // Send ID of the peer we want to talk to via the WebSocket connection
+    fn setup_call(&self, peer_id: &str) -> Result<OwnedMessage, Error> {
+        println!("Setting up signalling server call with {}", peer_id);
+        Ok(OwnedMessage::Text(format!("SESSION {}", peer_id)))
+    }
+
+    // Once we got the HELLO message from the WebSocket connection, start setting up the call
+    fn handle_hello(&self) -> Result<Option<OwnedMessage>, Error> {
+        if let Some(ref peer_id) = self.0.peer_id {
+            self.setup_call(peer_id).map(Some)
+        } else {
+            // Wait for a peer to appear
+            Ok(None)
+        }
+    }
+
+    // Once the session is set up correctly we start our pipeline
+    fn handle_session_ok(&self) -> Result<Option<OwnedMessage>, Error> {
+        self.setup_pipeline()?;
+
         // And finally asynchronously start our pipeline
         let app_clone = self.downgrade();
         self.0.pipeline.call_async(move |pipeline| {
@@ -468,23 +535,7 @@ impl App {
             }
         });
 
-        Ok(())
+        Ok(None)
-    }
-
-    // Send ID of the peer we want to talk to via the WebSocket connection
-    fn setup_call(&self) -> Result<OwnedMessage, Error> {
-        println!("Setting up signalling server call with {}", self.0.peer_id);
-        Ok(OwnedMessage::Text(format!("SESSION {}", &self.0.peer_id)))
-    }
-
-    // Once we got the HELLO message from the WebSocket connection, start setting up the call
-    fn handle_hello(&self) -> Result<Option<OwnedMessage>, Error> {
-        self.setup_call().map(Some)
-    }
-
-    // Once the session is set up correctly we start our pipeline
-    fn handle_session_ok(&self) -> Result<Option<OwnedMessage>, Error> {
-        self.start_pipeline().map(|_| None)
     }
 
     // Handle errors from the peer send to us via the WebSocket connection
@@ -495,16 +546,12 @@ impl App {
     }
 
     // Handle incoming SDP answers from the peer
-    //
-    // TODO: We currently only handle answers, not offers
     fn handle_sdp(&self, type_: &str, sdp: &str) -> Result<Option<OwnedMessage>, Error> {
-        if type_ != "answer" {
+        if type_ == "answer" {
-            return Err(PeerError("Sdp type is not \"answer\"".into()).into());
-        }
-
             print!("Received answer:\n{}\n", sdp);
 
-        let ret = gst_sdp::SDPMessage::parse_buffer(sdp.as_bytes()).unwrap();
+            let ret = gst_sdp::SDPMessage::parse_buffer(sdp.as_bytes())
+                .map_err(|_| GStreamerError("Failed to parse SDP answer".into()))?;
             let answer =
                 gst_webrtc::WebRTCSessionDescription::new(gst_webrtc::WebRTCSDPType::Answer, ret);
             self.0
@@ -513,6 +560,57 @@ impl App {
                 .unwrap();
 
             Ok(None)
+        } else if type_ == "offer" {
+            print!("Received offer:\n{}\n", sdp);
+
+            // Need to start the pipeline as a first step here
+            self.setup_pipeline()?;
+
+            let ret = gst_sdp::SDPMessage::parse_buffer(sdp.as_bytes())
+                .map_err(|_| GStreamerError("Failed to parse SDP offer".into()))?;
+
+            // And then asynchronously start our pipeline and do the next steps. The
+            // pipeline needs to be started before we can create an answer
+            let app_clone = self.downgrade();
+            self.0.pipeline.call_async(move |pipeline| {
+                let app = upgrade_weak!(app_clone);
+
+                if let Err(err) = pipeline.set_state(gst::State::Playing) {
+                    app.post_error(
+                        format!("Failed to set pipeline to Playing: {:?}", err).as_str(),
+                    );
+                    return;
+                }
+
+                let offer = gst_webrtc::WebRTCSessionDescription::new(
+                    gst_webrtc::WebRTCSDPType::Offer,
+                    ret,
+                );
+
+                app.0
+                    .webrtcbin
+                    .emit("set-remote-description", &[&offer, &None::<gst::Promise>])
+                    .unwrap();
+
+                let app_clone = app.downgrade();
+                let promise = gst::Promise::new_with_change_func(move |promise| {
+                    let app = upgrade_weak!(app_clone);
+
+                    if let Err(err) = app.on_answer_created(promise) {
+                        app.post_error(format!("Failed to send SDP answer: {:?}", err).as_str());
+                    }
+                });
+
+                app.0
+                    .webrtcbin
+                    .emit("create-answer", &[&None::<gst::Structure>, &promise])
+                    .unwrap();
+            });
+
+            Ok(None)
+        } else {
+            Err(PeerError(format!("Sdp type is not \"answer\" but \"{}\"", type_)).into())
+        }
     }
 
     // Handle incoming ICE candidates from the peer by passing them to webrtcbin
@@ -601,13 +699,13 @@ impl App {
     }
 }
 
-fn parse_args() -> (String, String) {
+fn parse_args() -> (String, Option<String>) {
     let matches = clap::App::new("Sendrecv rust")
         .arg(
             clap::Arg::with_name("peer-id")
                 .help("String ID of the peer to connect to")
                 .long("peer-id")
-                .required(true)
+                .required(false)
                 .takes_value(true),
         )
         .arg(
@@ -623,9 +721,9 @@ fn parse_args() -> (String, String) {
         .value_of("server")
         .unwrap_or("wss://webrtc.nirbheek.in:8443");
 
-    let peer_id = matches.value_of("peer-id").unwrap();
+    let peer_id = matches.value_of("peer-id");
 
-    (server.to_string(), peer_id.to_string())
+    (server.to_string(), peer_id.map(String::from))
 }
 
 fn check_plugins() -> Result<(), Error> {
@@ -696,7 +794,7 @@ fn main() {
                 // Create our application control logic
                 let (send_ws_msg_tx, send_ws_msg_rx) = mpsc::unbounded_channel::<OwnedMessage>();
                 let app = App(Arc::new(AppInner {
-                    peer_id: peer_id.to_string(),
+                    peer_id,
                     pipeline,
                     webrtcbin,
                     send_msg_tx: Mutex::new(send_ws_msg_tx),
@@ -721,7 +819,7 @@ fn main() {
                 // and convert them into potential messages to send out
                 let app_clone = app.clone();
                 let gst_messages = send_gst_msg_rx
-                    .map_err(|err| Error::from(err))
+                    .map_err(Error::from)
                     .and_then(move |msg| app_clone.handle_pipeline_message(&msg))
                     .filter_map(|msg| msg);
 
@@ -730,7 +828,7 @@ fn main() {
 
                 // And here collect all the asynchronous outgoing messages that come
                 // from other threads
-                let async_outgoing_messages = send_ws_msg_rx.map_err(|err| Error::from(err));
+                let async_outgoing_messages = send_ws_msg_rx.map_err(Error::from);
 
                 // Merge both outgoing messages streams and send them out directly
                 sink.sink_map_err(|err| Error::from(WebSocketError(err)))