Files
webrtc/examples/simulcast/main.go
Sean DuBois fe447d6e56 Revert "Process RTCP Packets in OnTrack examples"
This is not needed. We don't perform any operations on inbound RTCP
packets. Receiver Reports and TWCC are generated by Reading RTP packets.

This reverts commit 080d7b8427.
2021-12-29 23:39:32 -05:00

190 lines
5.7 KiB
Go

// +build !js
package main
import (
"errors"
"fmt"
"io"
"os"
"time"
"github.com/pion/interceptor"
"github.com/pion/rtcp"
"github.com/pion/webrtc/v3"
"github.com/pion/webrtc/v3/examples/internal/signal"
)
func main() {
// Everything below is the Pion WebRTC API! Thanks for using it ❤️.
// Prepare the configuration
config := webrtc.Configuration{
ICEServers: []webrtc.ICEServer{
{
URLs: []string{"stun:stun.l.google.com:19302"},
},
},
}
// Enable Extension Headers needed for Simulcast
m := &webrtc.MediaEngine{}
if err := m.RegisterDefaultCodecs(); err != nil {
panic(err)
}
for _, extension := range []string{
"urn:ietf:params:rtp-hdrext:sdes:mid",
"urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id",
"urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id",
} {
if err := m.RegisterHeaderExtension(webrtc.RTPHeaderExtensionCapability{URI: extension}, webrtc.RTPCodecTypeVideo); err != nil {
panic(err)
}
}
// Create a InterceptorRegistry. This is the user configurable RTP/RTCP Pipeline.
// This provides NACKs, RTCP Reports and other features. If you use `webrtc.NewPeerConnection`
// this is enabled by default. If you are manually managing You MUST create a InterceptorRegistry
// for each PeerConnection.
i := &interceptor.Registry{}
// Use the default set of Interceptors
if err := webrtc.RegisterDefaultInterceptors(m, i); err != nil {
panic(err)
}
// Create a new RTCPeerConnection
peerConnection, err := webrtc.NewAPI(webrtc.WithMediaEngine(m), webrtc.WithInterceptorRegistry(i)).NewPeerConnection(config)
if err != nil {
panic(err)
}
defer func() {
if cErr := peerConnection.Close(); cErr != nil {
fmt.Printf("cannot close peerConnection: %v\n", cErr)
}
}()
outputTracks := map[string]*webrtc.TrackLocalStaticRTP{}
// Create Track that we send video back to browser on
outputTrack, err := webrtc.NewTrackLocalStaticRTP(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeVP8}, "video_q", "pion_q")
if err != nil {
panic(err)
}
outputTracks["q"] = outputTrack
outputTrack, err = webrtc.NewTrackLocalStaticRTP(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeVP8}, "video_h", "pion_h")
if err != nil {
panic(err)
}
outputTracks["h"] = outputTrack
outputTrack, err = webrtc.NewTrackLocalStaticRTP(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeVP8}, "video_f", "pion_f")
if err != nil {
panic(err)
}
outputTracks["f"] = outputTrack
// Add this newly created track to the PeerConnection
if _, err = peerConnection.AddTrack(outputTracks["q"]); err != nil {
panic(err)
}
if _, err = peerConnection.AddTrack(outputTracks["h"]); err != nil {
panic(err)
}
if _, err = peerConnection.AddTrack(outputTracks["f"]); err != nil {
panic(err)
}
// Read incoming RTCP packets
// Before these packets are returned they are processed by interceptors. For things
// like NACK this needs to be called.
processRTCP := func(rtpSender *webrtc.RTPSender) {
rtcpBuf := make([]byte, 1500)
for {
if _, _, rtcpErr := rtpSender.Read(rtcpBuf); rtcpErr != nil {
return
}
}
}
for _, rtpSender := range peerConnection.GetSenders() {
go processRTCP(rtpSender)
}
// Wait for the offer to be pasted
offer := webrtc.SessionDescription{}
signal.Decode(signal.MustReadStdin(), &offer)
if err = peerConnection.SetRemoteDescription(offer); err != nil {
panic(err)
}
// Set a handler for when a new remote track starts
peerConnection.OnTrack(func(track *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {
fmt.Println("Track has started")
// Start reading from all the streams and sending them to the related output track
rid := track.RID()
go func() {
ticker := time.NewTicker(3 * time.Second)
for range ticker.C {
fmt.Printf("Sending pli for stream with rid: %q, ssrc: %d\n", track.RID(), track.SSRC())
if writeErr := peerConnection.WriteRTCP([]rtcp.Packet{&rtcp.PictureLossIndication{MediaSSRC: uint32(track.SSRC())}}); writeErr != nil {
fmt.Println(writeErr)
}
}
}()
for {
// Read RTP packets being sent to Pion
packet, _, readErr := track.ReadRTP()
if readErr != nil {
panic(readErr)
}
if writeErr := outputTracks[rid].WriteRTP(packet); writeErr != nil && !errors.Is(writeErr, io.ErrClosedPipe) {
panic(writeErr)
}
}
})
// Set the handler for Peer connection state
// This will notify you when the peer has connected/disconnected
peerConnection.OnConnectionStateChange(func(s webrtc.PeerConnectionState) {
fmt.Printf("Peer Connection State has changed: %s\n", s.String())
if s == webrtc.PeerConnectionStateFailed {
// Wait until PeerConnection has had no network activity for 30 seconds or another failure. It may be reconnected using an ICE Restart.
// Use webrtc.PeerConnectionStateDisconnected if you are interested in detecting faster timeout.
// Note that the PeerConnection may come back from PeerConnectionStateDisconnected.
fmt.Println("Peer Connection has gone to failed exiting")
os.Exit(0)
}
})
// Create an answer
answer, err := peerConnection.CreateAnswer(nil)
if err != nil {
panic(err)
}
// Create channel that is blocked until ICE Gathering is complete
gatherComplete := webrtc.GatheringCompletePromise(peerConnection)
// Sets the LocalDescription, and starts our UDP listeners
err = peerConnection.SetLocalDescription(answer)
if err != nil {
panic(err)
}
// Block until ICE Gathering is complete, disabling trickle ICE
// we do this because we only can exchange one signaling message
// in a production application you should exchange ICE Candidates via OnICECandidate
<-gatherComplete
// Output the answer in base64 so we can paste it in browser
fmt.Println(signal.Encode(*peerConnection.LocalDescription()))
// Block forever
select {}
}