api: add more attributes to WebRTC connections

new attributes: peerConnectionEstablished, localCandidate, remoteCandidate

apidocs/openapi.yaml

@@ -534,6 +534,12 @@ components:
           type: string
         remoteAddr:
           type: string
+        peerConnectionEstablished:
+          type: bool
+        localCandidate:
+          type: string
+        remoteCandidate:
+          type: string
         bytesReceived:
           type: number
         bytesSent:

internal/core/api_test.go

@@ -405,6 +405,7 @@ func TestAPIProtocolSpecificList(t *testing.T) {
 		"rtmp",
 		"rtmps",
 		"hls",
+		"webrtc",
 	} {
 		t.Run(ca, func(t *testing.T) {
 			conf := "api: yes\n"
@@ -490,7 +491,6 @@ func TestAPIProtocolSpecificList(t *testing.T) {
 
 			case "hls":
 				source := gortsplib.Client{}
-
 				err := source.StartRecording("rtsp://localhost:8554/mypath",
 					media.Medias{medi})
 				require.NoError(t, err)
@@ -502,6 +502,17 @@ func TestAPIProtocolSpecificList(t *testing.T) {
 					defer res.Body.Close()
 					require.Equal(t, 200, res.StatusCode)
 				}()
+
+			case "webrtc":
+				source := gortsplib.Client{}
+				err := source.StartRecording("rtsp://localhost:8554/mypath",
+					media.Medias{medi})
+				require.NoError(t, err)
+				defer source.Close()
+
+				c, err := newWebRTCTestClient("ws://localhost:8889/mypath/ws")
+				require.NoError(t, err)
+				defer c.close()
 			}
 
 			switch ca {
@@ -562,6 +573,31 @@ func TestAPIProtocolSpecificList(t *testing.T) {
 				s := fmt.Sprintf("^%d-", time.Now().Year())
 				require.Regexp(t, s, out.Items[firstID].Created)
 				require.Regexp(t, s, out.Items[firstID].LastRequest)
+
+			case "webrtc":
+				type item struct {
+					Created                   time.Time `json:"created"`
+					RemoteAddr                string    `json:"remoteAddr"`
+					PeerConnectionEstablished bool      `json:"peerConnectionEstablished"`
+					LocalCandidate            string    `json:"localCandidate"`
+					RemoteCandidate           string    `json:"remoteCandidate"`
+					BytesReceived             uint64    `json:"bytesReceived"`
+					BytesSent                 uint64    `json:"bytesSent"`
+				}
+
+				var out struct {
+					Items map[string]item `json:"items"`
+				}
+				err = httpRequest(http.MethodGet, "http://localhost:9997/v1/webrtcconns/list", nil, &out)
+				require.NoError(t, err)
+
+				var firstID string
+				for k := range out.Items {
+					firstID = k
+				}
+
+				itm := out.Items[firstID]
+				require.Equal(t, true, itm.PeerConnectionEstablished)
 			}
 		})
 	}

internal/core/core.go

@@ -565,7 +565,7 @@ func (p *Core) closeResources(newConf *conf.Conf, calledByAPI bool) {
 		closePathManager ||
 		closeMetrics
 
-	closeWebrtcServer := newConf == nil ||
+	closeWebRTCServer := newConf == nil ||
 		newConf.WebRTCDisable != p.conf.WebRTCDisable ||
 		newConf.ExternalAuthenticationURL != p.conf.ExternalAuthenticationURL ||
 		newConf.WebRTCAddress != p.conf.WebRTCAddress ||
@@ -590,7 +590,7 @@ func (p *Core) closeResources(newConf *conf.Conf, calledByAPI bool) {
 		closeRTSPSServer ||
 		closeRTMPServer ||
 		closeHLSServer ||
-		closeWebrtcServer
+		closeWebRTCServer
 
 	if newConf == nil && p.confWatcher != nil {
 		p.confWatcher.Close()
@@ -621,7 +621,7 @@ func (p *Core) closeResources(newConf *conf.Conf, calledByAPI bool) {
 		p.pathManager = nil
 	}
 
-	if closeWebrtcServer && p.webRTCServer != nil {
+	if closeWebRTCServer && p.webRTCServer != nil {
 		p.webRTCServer.close()
 		p.webRTCServer = nil
 	}

internal/core/webrtc_conn.go

@@ -62,37 +62,6 @@ func newPeerConnection(configuration webrtc.Configuration,
 	return api.NewPeerConnection(configuration)
 }
 
-func describeActiveCandidates(pc *webrtc.PeerConnection) (string, string) {
-	var lcid string
-	var rcid string
-
-	for _, stats := range pc.GetStats() {
-		if tstats, ok := stats.(webrtc.ICECandidatePairStats); ok && tstats.Nominated {
-			lcid = tstats.LocalCandidateID
-			rcid = tstats.RemoteCandidateID
-			break
-		}
-	}
-
-	var ldesc string
-	var rdesc string
-
-	for _, stats := range pc.GetStats() {
-		if tstats, ok := stats.(webrtc.ICECandidateStats); ok {
-			str := tstats.CandidateType.String() + "/" + tstats.Protocol + "/" +
-				tstats.IP + "/" + strconv.FormatInt(int64(tstats.Port), 10)
-
-			if tstats.ID == lcid {
-				ldesc = str
-			} else if tstats.ID == rcid {
-				rdesc = str
-			}
-		}
-	}
-
-	return ldesc, rdesc
-}
-
 type webRTCTrack struct {
 	media       *media.Media
 	format      format.Format
@@ -187,13 +156,73 @@ func (c *webRTCConn) remoteAddr() net.Addr {
 	return c.wsconn.RemoteAddr()
 }
 
+func (c *webRTCConn) peerConnectionEstablished() bool {
+	c.mutex.RLock()
+	defer c.mutex.RUnlock()
+
+	return c.curPC != nil
+}
+
+func (c *webRTCConn) localCandidate() string {
+	c.mutex.RLock()
+	defer c.mutex.RUnlock()
+
+	if c.curPC != nil {
+		var cid string
+		for _, stats := range c.curPC.GetStats() {
+			if tstats, ok := stats.(webrtc.ICECandidatePairStats); ok && tstats.Nominated {
+				cid = tstats.LocalCandidateID
+				break
+			}
+		}
+
+		if cid != "" {
+			for _, stats := range c.curPC.GetStats() {
+				if tstats, ok := stats.(webrtc.ICECandidateStats); ok && tstats.ID == cid {
+					return tstats.CandidateType.String() + "/" + tstats.Protocol + "/" +
+						tstats.IP + "/" + strconv.FormatInt(int64(tstats.Port), 10)
+				}
+			}
+		}
+	}
+	return ""
+}
+
+func (c *webRTCConn) remoteCandidate() string {
+	c.mutex.RLock()
+	defer c.mutex.RUnlock()
+
+	if c.curPC != nil {
+		var cid string
+		for _, stats := range c.curPC.GetStats() {
+			if tstats, ok := stats.(webrtc.ICECandidatePairStats); ok && tstats.Nominated {
+				cid = tstats.RemoteCandidateID
+				break
+			}
+		}
+
+		if cid != "" {
+			for _, stats := range c.curPC.GetStats() {
+				if tstats, ok := stats.(webrtc.ICECandidateStats); ok && tstats.ID == cid {
+					return tstats.CandidateType.String() + "/" + tstats.Protocol + "/" +
+						tstats.IP + "/" + strconv.FormatInt(int64(tstats.Port), 10)
+				}
+			}
+		}
+	}
+	return ""
+}
+
 func (c *webRTCConn) bytesReceived() uint64 {
 	c.mutex.RLock()
 	defer c.mutex.RUnlock()
-	for _, stats := range c.curPC.GetStats() {
-		if tstats, ok := stats.(webrtc.TransportStats); ok {
-			if tstats.ID == "iceTransport" {
-				return tstats.BytesReceived
+
+	if c.curPC != nil {
+		for _, stats := range c.curPC.GetStats() {
+			if tstats, ok := stats.(webrtc.TransportStats); ok {
+				if tstats.ID == "iceTransport" {
+					return tstats.BytesReceived
+				}
 			}
 		}
 	}
@@ -203,10 +232,13 @@ func (c *webRTCConn) bytesReceived() uint64 {
 func (c *webRTCConn) bytesSent() uint64 {
 	c.mutex.RLock()
 	defer c.mutex.RUnlock()
-	for _, stats := range c.curPC.GetStats() {
-		if tstats, ok := stats.(webrtc.TransportStats); ok {
-			if tstats.ID == "iceTransport" {
-				return tstats.BytesSent
+
+	if c.curPC != nil {
+		for _, stats := range c.curPC.GetStats() {
+			if tstats, ok := stats.(webrtc.TransportStats); ok {
+				if tstats.ID == "iceTransport" {
+					return tstats.BytesSent
+				}
 			}
 		}
 	}
@@ -335,10 +367,6 @@ func (c *webRTCConn) runInner(ctx context.Context) error {
 		<-pcClosed
 	}()
 
-	c.mutex.Lock()
-	c.curPC = pc
-	c.mutex.Unlock()
-
 	for _, track := range tracks {
 		rtpSender, err := pc.AddTrack(track.webRTCTrack)
 		if err != nil {
@@ -440,8 +468,12 @@ outer:
 	// in order to allow the other side of the connection
 	// to switch to the "connected" state before WebSocket is closed.
 
-	ldesc, rdesc := describeActiveCandidates(pc)
-	c.log(logger.Info, "peer connection established, local candidate: %v, remote candidate: %v", ldesc, rdesc)
+	c.mutex.Lock()
+	c.curPC = pc
+	c.mutex.Unlock()
+
+	c.log(logger.Info, "peer connection established, local candidate: %v, remote candidate: %v",
+		c.localCandidate(), c.remoteCandidate())
 
 	ringBuffer, _ := ringbuffer.New(uint64(c.readBufferCount))
 	defer ringBuffer.Close()

internal/core/webrtc_server.go

@@ -32,10 +32,13 @@ var upgrader = websocket.Upgrader{
 }
 
 type webRTCServerAPIConnsListItem struct {
-	Created       time.Time `json:"created"`
-	RemoteAddr    string    `json:"remoteAddr"`
-	BytesReceived uint64    `json:"bytesReceived"`
-	BytesSent     uint64    `json:"bytesSent"`
+	Created                   time.Time `json:"created"`
+	RemoteAddr                string    `json:"remoteAddr"`
+	PeerConnectionEstablished bool      `json:"peerConnectionEstablished"`
+	LocalCandidate            string    `json:"localCandidate"`
+	RemoteCandidate           string    `json:"remoteCandidate"`
+	BytesReceived             uint64    `json:"bytesReceived"`
+	BytesSent                 uint64    `json:"bytesSent"`
 }
 
 type webRTCServerAPIConnsListData struct {
@@ -264,10 +267,13 @@ outer:
 
 			for c := range s.conns {
 				data.Items[c.uuid.String()] = webRTCServerAPIConnsListItem{
-					Created:       c.created,
-					RemoteAddr:    c.remoteAddr().String(),
-					BytesReceived: c.bytesReceived(),
-					BytesSent:     c.bytesSent(),
+					Created:                   c.created,
+					RemoteAddr:                c.remoteAddr().String(),
+					PeerConnectionEstablished: c.peerConnectionEstablished(),
+					LocalCandidate:            c.localCandidate(),
+					RemoteCandidate:           c.remoteCandidate(),
+					BytesReceived:             c.bytesReceived(),
+					BytesSent:                 c.bytesSent(),
 				}
 			}
 

internal/core/webrtc_server_test.go

@@ -14,49 +14,43 @@ import (
 	"github.com/stretchr/testify/require"
 )
 
-func TestWebRTCServer(t *testing.T) {
-	p, ok := newInstance("paths:\n" +
-		"  all:\n")
-	require.Equal(t, true, ok)
-	defer p.Close()
+type webRTCTestClient struct {
+	wc    *websocket.Conn
+	pc    *webrtc.PeerConnection
+	track chan *webrtc.TrackRemote
+}
 
-	medi := &media.Media{
-		Type: media.TypeVideo,
-		Formats: []format.Format{&format.H264{
-			PayloadTyp:        96,
-			PacketizationMode: 1,
-		}},
+func newWebRTCTestClient(addr string) (*webRTCTestClient, error) {
+	wc, _, err := websocket.DefaultDialer.Dial(addr, nil) //nolint:bodyclose
+	if err != nil {
+		return nil, err
 	}
 
-	v := gortsplib.TransportTCP
-	source := gortsplib.Client{
-		Transport: &v,
+	_, msg, err := wc.ReadMessage()
+	if err != nil {
+		wc.Close()
+		return nil, err
 	}
-	err := source.StartRecording("rtsp://localhost:8554/stream", media.Medias{medi})
-	require.NoError(t, err)
-	defer source.Close()
-
-	c, _, err := websocket.DefaultDialer.Dial("ws://localhost:8889/stream/ws", nil) //nolint:bodyclose
-	require.NoError(t, err)
-	defer c.Close()
-
-	_, msg, err := c.ReadMessage()
-	require.NoError(t, err)
 
 	var iceServers []webrtc.ICEServer
 	err = json.Unmarshal(msg, &iceServers)
-	require.NoError(t, err)
+	if err != nil {
+		wc.Close()
+		return nil, err
+	}
 
 	pc, err := newPeerConnection(webrtc.Configuration{
 		ICEServers: iceServers,
 	})
-	require.NoError(t, err)
-	defer pc.Close()
+	if err != nil {
+		wc.Close()
+		return nil, err
+	}
 
 	pc.OnICECandidate(func(i *webrtc.ICECandidate) {
 		if i != nil {
 			enc, _ := json.Marshal(i.ToJSON())
-			c.WriteMessage(websocket.TextMessage, enc)
+			wc.WriteMessage(websocket.TextMessage, enc)
 		}
 	})
 
@@ -67,39 +61,71 @@ func TestWebRTCServer(t *testing.T) {
 		}
 	})
 
-	track := make(chan *webrtc.TrackRemote)
+	track := make(chan *webrtc.TrackRemote, 1)
 	pc.OnTrack(func(trak *webrtc.TrackRemote, recv *webrtc.RTPReceiver) {
 		track <- trak
 	})
 
 	_, err = pc.AddTransceiverFromKind(webrtc.RTPCodecTypeVideo)
-	require.NoError(t, err)
+	if err != nil {
+		wc.Close()
+		pc.Close()
+		return nil, err
+	}
 
 	localOffer, err := pc.CreateOffer(nil)
-	require.NoError(t, err)
+	if err != nil {
+		wc.Close()
+		pc.Close()
+		return nil, err
+	}
 
 	enc, err := json.Marshal(localOffer)
-	require.NoError(t, err)
+	if err != nil {
+		wc.Close()
+		pc.Close()
+		return nil, err
+	}
 
-	err = c.WriteMessage(websocket.TextMessage, enc)
-	require.NoError(t, err)
+	err = wc.WriteMessage(websocket.TextMessage, enc)
+	if err != nil {
+		wc.Close()
+		pc.Close()
+		return nil, err
+	}
 
 	err = pc.SetLocalDescription(localOffer)
-	require.NoError(t, err)
+	if err != nil {
+		wc.Close()
+		pc.Close()
+		return nil, err
+	}
 
-	_, msg, err = c.ReadMessage()
-	require.NoError(t, err)
+	_, msg, err = wc.ReadMessage()
+	if err != nil {
+		wc.Close()
+		pc.Close()
+		return nil, err
+	}
 
 	var remoteOffer webrtc.SessionDescription
 	err = json.Unmarshal(msg, &remoteOffer)
-	require.NoError(t, err)
+	if err != nil {
+		wc.Close()
+		pc.Close()
+		return nil, err
+	}
 
 	err = pc.SetRemoteDescription(remoteOffer)
-	require.NoError(t, err)
+	if err != nil {
+		wc.Close()
+		pc.Close()
+		return nil, err
+	}
 
 	go func() {
 		for {
-			_, msg, err := c.ReadMessage()
+			_, msg, err := wc.ReadMessage()
 			if err != nil {
 				return
 			}
@@ -116,6 +142,44 @@ func TestWebRTCServer(t *testing.T) {
 
 	<-connected
 
+	return &webRTCTestClient{
+		wc:    wc,
+		pc:    pc,
+		track: track,
+	}, nil
+}
+
+func (c *webRTCTestClient) close() {
+	c.pc.Close()
+	c.wc.Close()
+}
+
+func TestWebRTCServer(t *testing.T) {
+	p, ok := newInstance("paths:\n" +
+		"  all:\n")
+	require.Equal(t, true, ok)
+	defer p.Close()
+
+	medi := &media.Media{
+		Type: media.TypeVideo,
+		Formats: []format.Format{&format.H264{
+			PayloadTyp:        96,
+			PacketizationMode: 1,
+		}},
+	}
+
+	v := gortsplib.TransportTCP
+	source := gortsplib.Client{
+		Transport: &v,
+	}
+	err := source.StartRecording("rtsp://localhost:8554/stream", media.Medias{medi})
+	require.NoError(t, err)
+	defer source.Close()
+
+	c, err := newWebRTCTestClient("ws://localhost:8889/stream/ws")
+	require.NoError(t, err)
+	defer c.close()
+
 	time.Sleep(500 * time.Millisecond)
 
 	source.WritePacketRTP(medi, &rtp.Packet{
@@ -130,7 +194,7 @@ func TestWebRTCServer(t *testing.T) {
 		Payload: []byte{0x01, 0x02, 0x03, 0x04},
 	})
 
-	trak := <-track
+	trak := <-c.track
 
 	pkt, _, err := trak.ReadRTP()
 	require.NoError(t, err)