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This commit is contained in:
gbrian
2021-03-29 11:03:25 +00:00
parent a1fcf87345
commit 8efc5d7094
95 changed files with 5789 additions and 874 deletions
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2
server/internal/webrtc/handle.go
View File

@ -5,7 +5,7 @@ import (
"encoding/binary"
"strconv"
"github.com/pion/webrtc/v2"
"github.com/pion/webrtc/v3"
)
const OP_MOVE = 0x01

4
server/internal/webrtc/peer.go
View File

@ -3,7 +3,7 @@ package webrtc
import (
"sync"
"github.com/pion/webrtc/v2"
"github.com/pion/webrtc/v3"
)
type Peer struct {
@ -28,7 +28,7 @@ func (peer *Peer) WriteData(v interface{}) error {
}
func (peer *Peer) Destroy() error {
if peer.connection != nil && peer.connection.ConnectionState() == webrtc.PeerConnectionStateConnected {
if peer.connection != nil && peer.connection.ConnectionState() != webrtc.PeerConnectionStateClosed {
if err := peer.connection.Close(); err != nil {
return err
}

162
server/internal/webrtc/webrtc.go
View File

@ -1,13 +1,15 @@
package webrtc
import (
"encoding/json"
"fmt"
"io"
"math/rand"
"strings"
"time"
"github.com/pion/webrtc/v2"
"github.com/pion/webrtc/v2/pkg/media"
"github.com/pion/interceptor"
"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
@ -97,15 +99,22 @@ 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.SetICETimeouts(6 * time.Second, 6 * time.Second, 3 * time.Second)
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)
@ -113,19 +122,10 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
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)
negotiated := true
_, err = connection.CreateDataChannel("data", &webrtc.DataChannelInit{
Negotiated: &negotiated,
})
if err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
@ -138,21 +138,69 @@ 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) {
manager.logger.Info().
Str("connection_state", connectionState.String()).
Msg("connection state has changed")
})
rtpVideo, err := connection.AddTrack(manager.videoTrack);
if err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
rtpAudio, err := connection.AddTrack(manager.audioTrack);
if 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 {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
connection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
switch state {
case webrtc.PeerConnectionStateDisconnected:
case webrtc.PeerConnectionStateFailed:
manager.logger.Info().Str("id", id).Msg("peer disconnected")
manager.sessions.Destroy(id)
break
case webrtc.PeerConnectionStateFailed:
manager.logger.Warn().Str("id", id).Msg("peer failed")
manager.sessions.Destroy(id)
case webrtc.PeerConnectionStateClosed:
manager.logger.Info().Str("id", id).Msg("peer closed")
manager.sessions.Destroy(id)
case webrtc.PeerConnectionStateConnected:
manager.logger.Info().Str("id", id).Msg("peer connected")
if err = session.SetConnected(true); err != nil {
manager.logger.Warn().Err(err).Msg("unable to set connected on peer")
manager.sessions.Destroy(id)
}
break
}
})
connection.OnICECandidate(func(i *webrtc.ICECandidate) {
if i == nil {
manager.logger.Info().Msg("sent all ICECandidates")
return
}
candidateString, err := json.Marshal(i.ToJSON())
if err != nil {
manager.logger.Warn().Err(err).Msg("converting ICECandidate to json failed")
return
}
if err := session.SignalCandidate(string(candidateString)); err != nil {
manager.logger.Warn().Err(err).Msg("sending SignalCandidate failed")
return
}
})
@ -168,33 +216,55 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
return "", manager.config.ICELite, manager.config.ICEServers, err
}
go func() {
rtcpBuf := make([]byte, 1500)
for {
if _, _, rtcpErr := rtpVideo.Read(rtcpBuf); rtcpErr != nil {
return
}
}
}()
go func() {
rtcpBuf := make([]byte, 1500)
for {
if _, _, rtcpErr := rtpAudio.Read(rtcpBuf); rtcpErr != nil {
return
}
}
}()
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
fb := []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: fb}, PayloadType: 111}
case "G722":
codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/G722", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 9}
case "PCMU":
codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/PCMU", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 0}
case "PCMA":
codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/PCMA", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, 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