update to pion v3

This commit is contained in:
Marcel Battista 2021-02-14 16:30:24 +00:00
parent 00a785f4c5
commit a362df4976
14 changed files with 211 additions and 84 deletions

View file

@ -1,13 +1,15 @@
package webrtc
import (
"encoding/json"
"fmt"
"io"
"math/rand"
"strings"
"github.com/pion/webrtc/v2"
"github.com/pion/webrtc/v2/pkg/media"
"github.com/pion/interceptor"
"github.com/pion/rtcp"
"github.com/pion/webrtc/v3"
"github.com/pion/webrtc/v3/pkg/media"
"github.com/rs/zerolog"
"github.com/rs/zerolog/log"
@ -26,10 +28,10 @@ func New(sessions types.SessionManager, remote types.RemoteManager, config *conf
type WebRTCManager struct {
logger zerolog.Logger
videoTrack *webrtc.Track
audioTrack *webrtc.Track
videoCodec *webrtc.RTPCodec
audioCodec *webrtc.RTPCodec
videoTrack *webrtc.TrackLocalStaticSample
audioTrack *webrtc.TrackLocalStaticSample
videoCodec webrtc.RTPCodecParameters
audioCodec webrtc.RTPCodecParameters
sessions types.SessionManager
remote types.RemoteManager
config *config.WebRTC
@ -97,39 +99,31 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
settings.SetEphemeralUDPPortRange(manager.config.EphemeralMin, manager.config.EphemeralMax)
settings.SetNAT1To1IPs(manager.config.NAT1To1IPs, webrtc.ICECandidateTypeHost)
settings.SetSRTPReplayProtectionWindow(512)
// Create MediaEngine based off sdp
engine := webrtc.MediaEngine{}
engine.RegisterCodec(manager.audioCodec)
engine.RegisterCodec(manager.videoCodec)
engine.RegisterCodec(manager.audioCodec, webrtc.RTPCodecTypeAudio)
engine.RegisterCodec(manager.videoCodec, webrtc.RTPCodecTypeVideo)
i := &interceptor.Registry{}
if err := webrtc.RegisterDefaultInterceptors(&engine, i); err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
// Create API with MediaEngine and SettingEngine
api := webrtc.NewAPI(webrtc.WithMediaEngine(engine), webrtc.WithSettingEngine(settings))
api := webrtc.NewAPI(webrtc.WithMediaEngine(&engine), webrtc.WithSettingEngine(settings), webrtc.WithInterceptorRegistry(i))
// Create new peer connection
connection, err := api.NewPeerConnection(*configuration)
if err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
if _, err = connection.AddTransceiverFromTrack(manager.videoTrack, webrtc.RtpTransceiverInit{
Direction: webrtc.RTPTransceiverDirectionSendonly,
}); err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
if _, err = connection.AddTransceiverFromTrack(manager.audioTrack, webrtc.RtpTransceiverInit{
Direction: webrtc.RTPTransceiverDirectionSendonly,
}); err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
description, err := connection.CreateOffer(nil)
if err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
negotiated := true
connection.CreateDataChannel("data", &webrtc.DataChannelInit{
Negotiated: &negotiated,
})
connection.OnDataChannel(func(d *webrtc.DataChannel) {
d.OnMessage(func(msg webrtc.DataChannelMessage) {
if err = manager.handle(id, msg); err != nil {
@ -138,7 +132,31 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
})
})
connection.SetLocalDescription(description)
// Set the handler for ICE connection state
// This will notify you when the peer has connected/disconnected
connection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
fmt.Printf("Connection State has changed %s \n", connectionState.String())
})
rtpSender, viderr := connection.AddTrack(manager.videoTrack)
if viderr != nil {
return "", manager.config.ICELite, manager.config.ICEServers, viderr
}
if _, err = connection.AddTrack(manager.audioTrack); err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
description, err := connection.CreateOffer(nil)
if err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
err = connection.SetLocalDescription(description)
if err != nil {
panic(err)
}
connection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
switch state {
case webrtc.PeerConnectionStateDisconnected:
@ -156,6 +174,47 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
}
})
connection.OnICECandidate(func(i *webrtc.ICECandidate) {
if i != nil {
candidateString, err := json.Marshal(i.ToJSON())
if err != nil {
manager.logger.Info().Msg("error")
return
}
if err = session.SignalCandidate(string(candidateString));err != nil {
manager.logger.Info().Msg("err")
return
}
}
})
// Read incoming RTCP packets
// Before these packets are retuned they are processed by interceptors. For things
// like NACK this needs to be called.
go func() {
rtcpBuf := make([]byte, 1500)
for {
n, _, rtcpErr := rtpSender.Read(rtcpBuf)
if rtcpErr != nil {
return
}
ps, err := rtcp.Unmarshal(rtcpBuf[:n])
if err != nil {
log.Printf("Unmarshal RTCP: %v", err)
continue
}
for _, p := range ps {
switch p.(type) {
case *rtcp.TransportLayerNack:
manager.logger.Info().Msg("got a nack")
}
}
}
}()
if err := session.SetPeer(&Peer{
id: id,
api: api,
@ -171,30 +230,40 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
return description.SDP, manager.config.ICELite, manager.config.ICEServers, nil
}
func (m *WebRTCManager) createTrack(codecName string) (*webrtc.Track, *webrtc.RTPCodec, error) {
var codec *webrtc.RTPCodec
func (m *WebRTCManager) createTrack(codecName string) (*webrtc.TrackLocalStaticSample, webrtc.RTPCodecParameters, error) {
var codec webrtc.RTPCodecParameters
var fb []webrtc.RTCPFeedback
var fba []webrtc.RTCPFeedback
fb = []webrtc.RTCPFeedback{
{"goog-remb", ""},
{"nack", ""},
{"nack", "pli"},
{"ccm", "fir"},
}
fba = []webrtc.RTCPFeedback{}
switch codecName {
case webrtc.VP8:
codec = webrtc.NewRTPVP8Codec(webrtc.DefaultPayloadTypeVP8, 90000)
case webrtc.VP9:
codec = webrtc.NewRTPVP9Codec(webrtc.DefaultPayloadTypeVP9, 90000)
case webrtc.H264:
codec = webrtc.NewRTPH264Codec(webrtc.DefaultPayloadTypeH264, 90000)
case webrtc.Opus:
codec = webrtc.NewRTPOpusCodec(webrtc.DefaultPayloadTypeOpus, 48000)
case webrtc.G722:
codec = webrtc.NewRTPG722Codec(webrtc.DefaultPayloadTypeG722, 8000)
case webrtc.PCMU:
codec = webrtc.NewRTPPCMUCodec(webrtc.DefaultPayloadTypePCMU, 8000)
case webrtc.PCMA:
codec = webrtc.NewRTPPCMACodec(webrtc.DefaultPayloadTypePCMA, 8000)
case "VP8":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/VP8", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 96,}
case "VP9":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/VP9", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 98,}
case "H264":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/H264", ClockRate: 90000, Channels: 0, SDPFmtpLine: "level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f", RTCPFeedback: fb}, PayloadType: 102,}
case "Opus":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/opus", ClockRate: 48000, Channels: 2, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 111,}
case "G722":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/G722", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 9,}
case "PCMU":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/PCMU", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 0,}
case "PCMA":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/PCMA", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 8,}
default:
return nil, nil, fmt.Errorf("unknown codec %s", codecName)
return nil, codec, fmt.Errorf("unknown codec %s", codecName)
}
track, err := webrtc.NewTrack(codec.PayloadType, rand.Uint32(), "stream", "stream", codec)
track, err := webrtc.NewTrackLocalStaticSample(codec.RTPCodecCapability, "stream", "stream")
if err != nil {
return nil, nil, err
return nil, codec, err
}
return track, codec, nil