provides cancelation when a component stop responding it closes all resources

FAQ.md

@@ -1,5 +1,18 @@
 # FAQ
 
+## I can't connect two tabs or browser at the same for the SRT
+
+It doesn't work When I try to connect in another browser or tab, or even when I try to refresh the current page. It raises an seemingly timeout error.
+
+```
+astisrt: connecting failed: astisrt: connecting failed: astisrt: Connection setup failure: connection timed out
+```
+
+Apparently both `ffmpeg` and `srt-live-transmit` won't allow multiple persistent connections.
+
+ref1 https://github.com/Haivision/srt/blob/master/docs/apps/srt-live-transmit.md#medium-srt
+ref2 https://github.com/asticode/go-astisrt/issues/6#issuecomment-1917076767
+
 ## It's not working on Firefox/Chrome/Edge.
 
 [WebRTC establishes a baseline set of codecs which all compliant browsers are required to support. Some browsers may choose to allow other codecs as well.](https://developer.mozilla.org/en-US/docs/Web/Media/Formats/WebRTC_codecs#supported_video_codecs)

internal/controllers/srt_controller.go

@@ -51,18 +51,15 @@ func NewSRTController(c *entities.Config, l *zap.Logger, lc fx.Lifecycle) (*SRTC
 	}, nil
 }
 
-func (c *SRTController) Connect(params *entities.RequestParams) (*astisrt.Connection, error) {
+func (c *SRTController) Connect(cancel context.CancelFunc, params entities.RequestParams) (*astisrt.Connection, error) {
 	c.l.Sugar().Infow("trying to connect srt")
-	if params == nil {
-		return nil, entities.ErrMissingRemoteOffer
-	}
 
 	if err := params.Valid(); err != nil {
 		return nil, err
 	}
 
 	c.l.Sugar().Infow("Connecting to SRT ",
-		"offer", params,
+		"offer", params.String(),
 	)
 
 	conn, err := astisrt.Dial(astisrt.DialOptions{
@@ -74,9 +71,10 @@ func (c *SRTController) Connect(params *entities.RequestParams) (*astisrt.Connec
 		},
 
 		OnDisconnect: func(conn *astisrt.Connection, err error) {
-			c.l.Sugar().Fatalw("Disconnected from SRT",
+			c.l.Sugar().Infow("Canceling SRT",
 				"error", err,
 			)
+			cancel()
 		},
 
 		Host: params.SRTHost,

internal/controllers/streaming_controller.go

@@ -1,7 +1,6 @@
 package controllers
 
 import (
-	"context"
 	"encoding/json"
 	"fmt"
 	"io"
@@ -28,59 +27,69 @@ func NewStreamingController(c *entities.Config, l *zap.Logger) *StreamingControl
 	}
 }
 
-func (c *StreamingController) Stream(srtConnection *astisrt.Connection, videoTrack *webrtc.TrackLocalStaticSample, metadataTrack *webrtc.DataChannel) {
+func (c *StreamingController) Stream(sp entities.StreamParameters) {
 	r, w := io.Pipe()
+
 	defer r.Close()
 	defer w.Close()
-	defer srtConnection.Close()
+	defer sp.SRTConnection.Close()
+	defer sp.WebRTCConn.Close()
+	defer sp.Cancel()
 
 	c.l.Sugar().Infow("start streaming")
 
 	// TODO: pick the proper transport? is it possible to get rtp instead?
-	go c.readFromSRTIntoWriterPipe(srtConnection, w)
+	go c.readFromSRTIntoWriterPipe(sp.SRTConnection, w)
 
-	dmx := astits.NewDemuxer(context.Background(), r)
+	// reading from reader pipe into mpeg-ts demuxer
+	dmx := astits.NewDemuxer(sp.Ctx, r)
 	eia608Reader := eia608.NewEIA608Reader()
 	h264PID := uint16(0)
 
-	// reading from reader pipe
 	for {
-		d, err := dmx.NextData()
+		select {
-		if err != nil {
+		case <-sp.Ctx.Done():
-			c.l.Sugar().Errorw("failed to demux mpeg ts",
+			c.l.Sugar().Errorw("stream was cancelled")
-				"error", err,
+			return
-			)
+		default:
-			break
+			d, err := dmx.NextData()
-		}
-
-		if d.PMT != nil {
-			h264PID = c.captureMediaInfoAndSendToWebRTC(d, metadataTrack, h264PID)
-			c.captureBitrateAndSendToWebRTC(d, metadataTrack)
-		}
-
-		if d.PID == h264PID && d.PES != nil {
-			if err = videoTrack.WriteSample(media.Sample{Data: d.PES.Data, Duration: time.Second / 30}); err != nil {
-				c.l.Sugar().Errorw("failed to write a sample mpeg ts to web rtc",
-					"error", err,
-				)
-				break
-			}
-			captions, err := eia608Reader.Parse(d.PES)
 			if err != nil {
-				c.l.Sugar().Errorw("failed to parse eia 608",
+				c.l.Sugar().Errorw("failed to demux mpeg ts",
 					"error", err,
 				)
 				break
 			}
-			if captions != "" {
+
-				captionsMsg, err := eia608.BuildCaptionsMessage(d.PES.Header.OptionalHeader.PTS, captions)
+			if d.PMT != nil {
-				if err != nil {
+				h264PID = c.captureMediaInfoAndSendToWebRTC(d, sp.MetadataTrack, h264PID)
-					c.l.Sugar().Errorw("failed to build captions message",
+				c.captureBitrateAndSendToWebRTC(d, sp.MetadataTrack)
+			}
+
+			if d.PID == h264PID && d.PES != nil {
+				// writing video from mpeg-ts into webrtc
+				if err = sp.VideoTrack.WriteSample(media.Sample{Data: d.PES.Data, Duration: time.Second / 30}); err != nil {
+					c.l.Sugar().Errorw("failed to write a sample mpeg ts to web rtc",
 						"error", err,
 					)
 					break
 				}
-				metadataTrack.SendText(captionsMsg)
+				captions, err := eia608Reader.Parse(d.PES)
+				if err != nil {
+					c.l.Sugar().Errorw("failed to parse eia 608",
+						"error", err,
+					)
+					break
+				}
+				if captions != "" {
+					captionsMsg, err := eia608.BuildCaptionsMessage(d.PES.Header.OptionalHeader.PTS, captions)
+					if err != nil {
+						c.l.Sugar().Errorw("failed to build captions message",
+							"error", err,
+						)
+						break
+					}
+					sp.MetadataTrack.SendText(captionsMsg)
+				}
 			}
 		}
 	}

internal/controllers/webrtc_controller.go

@@ -1,6 +1,7 @@
 package controllers
 
 import (
+	"context"
 	"net"
 
 	"github.com/flavioribeiro/donut/internal/entities"
@@ -27,7 +28,7 @@ func NewWebRTCController(
 	}
 }
 
-func (c *WebRTCController) CreatePeerConnection() (*webrtc.PeerConnection, error) {
+func (c *WebRTCController) CreatePeerConnection(cancel context.CancelFunc) (*webrtc.PeerConnection, error) {
 	c.l.Sugar().Infow("trying to set up web rtc conn")
 
 	peerConnectionConfiguration := webrtc.Configuration{}
@@ -48,6 +49,18 @@ func (c *WebRTCController) CreatePeerConnection() (*webrtc.PeerConnection, error
 	}
 
 	peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
+		finished := connectionState == webrtc.ICEConnectionStateClosed ||
+			connectionState == webrtc.ICEConnectionStateDisconnected ||
+			connectionState == webrtc.ICEConnectionStateCompleted ||
+			connectionState == webrtc.ICEConnectionStateFailed
+
+		if finished {
+			c.l.Sugar().Infow("Canceling webrtc",
+				"status", connectionState.String(),
+			)
+			cancel()
+		}
+
 		c.l.Sugar().Infow("OnICEConnectionStateChange",
 			"status", connectionState.String(),
 		)

internal/entities/entities.go

@@ -1,8 +1,10 @@
 package entities
 
 import (
+	"context"
 	"fmt"
 
+	astisrt "github.com/asticode/go-astisrt/pkg"
 	"github.com/pion/webrtc/v3"
 )
 
@@ -67,6 +69,15 @@ type Track struct {
 	Type TrackType
 }
 
+type StreamParameters struct {
+	WebRTCConn    *webrtc.PeerConnection
+	Cancel        context.CancelFunc
+	Ctx           context.Context
+	SRTConnection *astisrt.Connection
+	VideoTrack    *webrtc.TrackLocalStaticSample
+	MetadataTrack *webrtc.DataChannel
+}
+
 type Config struct {
 	HTTPPort int32  `required:"true" default:"8080"`
 	HTTPHost string `required:"true" default:"0.0.0.0"`

internal/web/handlers/signaling.go

@@ -1,6 +1,7 @@
 package handlers
 
 import (
+	"context"
 	"encoding/json"
 	"net/http"
 
@@ -53,7 +54,9 @@ func (h *SignalingHandler) ServeHTTP(w http.ResponseWriter, r *http.Request) err
 		return err
 	}
 
-	peer, err := h.webRTCController.CreatePeerConnection()
+	ctx, cancel := context.WithCancel(context.Background())
+
+	peer, err := h.webRTCController.CreatePeerConnection(cancel)
 	if err != nil {
 		h.l.Sugar().Errorw("error while setting up web rtc connection",
 			"error", err,
@@ -99,7 +102,7 @@ func (h *SignalingHandler) ServeHTTP(w http.ResponseWriter, r *http.Request) err
 		return err
 	}
 
-	srtConnection, err := h.srtController.Connect(&params)
+	srtConnection, err := h.srtController.Connect(cancel, params)
 	if err != nil {
 		h.l.Sugar().Errorw("error while connecting to an srt server",
 			"error", err,
@@ -107,7 +110,14 @@ func (h *SignalingHandler) ServeHTTP(w http.ResponseWriter, r *http.Request) err
 		return err
 	}
 
-	go h.streamingController.Stream(srtConnection, videoTrack, metadataSender)
+	go h.streamingController.Stream(entities.StreamParameters{
+		Cancel:        cancel,
+		Ctx:           ctx,
+		WebRTCConn:    peer,
+		SRTConnection: srtConnection,
+		VideoTrack:    videoTrack,
+		MetadataTrack: metadataSender,
+	})
 
 	w.Header().Set("Content-Type", "application/json")
 	w.WriteHeader(http.StatusOK)