Pion WebRTC- go语言实现的webrtc框架库

网友投稿 2401 2022-10-25

Pion WebRTC- go语言实现的webrtc框架库

Pion WebRTC- go语言实现的webrtc框架库

纯go语言实现的webrtc框架库。可以用于开发webrtc服务器和webrtc客户端。API接口与JavaScript类似,减少学习成本。

示例代码

package mainimport ( "fmt" "math/rand" "time" "github.com/pion/rtcp" "github.com/pion/webrtc/v2" "github.com/pion/webrtc/v2/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"}, }, }, } // Create a new RTCPeerConnection peerConnection, err := webrtc.NewPeerConnection(config) if err != nil { panic(err) } // Create Track that we send video back to browser on outputTrack, err := peerConnection.NewTrack(webrtc.DefaultPayloadTypeVP8, rand.Uint32(), "video", "pion") if err != nil { panic(err) } // Add this newly created track to the PeerConnection if _, err = peerConnection.AddTrack(outputTrack); err != nil { panic(err) } // Set a handler for when a new remote track starts, this handler copies inbound RTP packets, // replaces the SSRC and sends them back peerConnection.OnTrack(func(track *webrtc.Track, receiver *webrtc.RTPReceiver) { // Send a PLI on an interval so that the publisher is pushing a keyframe every rtcpPLIInterval // This is a temporary fix until we implement incoming RTCP events, then we would push a PLI only when a viewer requests it go func() { ticker := time.NewTicker(time.Second * 3) for range ticker.C { errSend := peerConnection.WriteRTCP([]rtcp.Packet{&rtcp.PictureLossIndication{MediaSSRC: track.SSRC()}}) if errSend != nil { fmt.Println(errSend) } } }() fmt.Printf("Track has started, of type %d: %s \n", track.PayloadType(), track.Codec().Name) for { // Read RTP packets being sent to Pion rtp, readErr := track.ReadRTP() if readErr != nil { panic(readErr) } // Replace the SSRC with the SSRC of the outbound track. // The only change we are making replacing the SSRC, the RTP packets are unchanged otherwise rtp.SSRC = outputTrack.SSRC() rtp.PayloadType = webrtc.DefaultPayloadTypeVP8 if writeErr := outputTrack.WriteRTP(rtp); writeErr != nil { panic(writeErr) } } }) // Set the handler for ICE connection state // This will notify you when the peer has connected/disconnected peerConnection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) { fmt.Printf("Connection State has changed %s \n", connectionState.String()) }) // Wait for the offer to be pasted offer := webrtc.SessionDescription{} signal.Decode(signal.MustReadStdin(), &offer) // Set the remote SessionDescription err = peerConnection.SetRemoteDescription(offer) if err != nil { panic(err) } // Create an answer answer, err := peerConnection.CreateAnswer(nil) if err != nil { panic(err) } // Sets the LocalDescription, and starts our UDP listeners err = peerConnection.SetLocalDescription(answer) if err != nil { panic(err) } // Output the answer in base64 so we can paste it in browser fmt.Println(signal.Encode(answer)) // Block forever select {}}

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