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fix logging for WebRTC.

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This commit is contained in:
m1k1o 2021-02-14 21:39:05 +01:00
parent 8ef91be6ad
commit 29fc67aff9
1 changed files with 41 additions and 31 deletions
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72
server/internal/webrtc/webrtc.go
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@ -120,10 +120,15 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
if err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
negotiated := true
connection.CreateDataChannel("data", &webrtc.DataChannelInit{
_, err = connection.CreateDataChannel("data", &webrtc.DataChannelInit{
Negotiated: &negotiated,
})
if err != nil {
return "", manager.config.ICELite, manager.config.ICEServers, err
}
connection.OnDataChannel(func(d *webrtc.DataChannel) {
d.OnMessage(func(msg webrtc.DataChannelMessage) {
if err = manager.handle(id, msg); err != nil {
@ -135,7 +140,9 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
// 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())
manager.logger.Info().
Str("connection_state", connectionState.String()).
Msg("connection state has changed")
})
rtpSender, viderr := connection.AddTrack(manager.videoTrack)
@ -154,7 +161,7 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
err = connection.SetLocalDescription(description)
if err != nil {
panic(err)
return "", manager.config.ICELite, manager.config.ICEServers, err
}
connection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
@ -175,46 +182,50 @@ 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 i == nil {
manager.logger.Info().Msg("sent all ICECandidates")
return
}
if err = session.SignalCandidate(string(candidateString));err != nil {
manager.logger.Info().Msg("err")
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
}
})
// 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 {
n, _, err := rtpSender.Read(rtcpBuf)
if err != nil {
return
}
ps, err := rtcp.Unmarshal(rtcpBuf[:n])
if err != nil {
log.Printf("Unmarshal RTCP: %v", err)
if err != nil {
manager.logger.Warn().Err(err).Msg("unmarshal RTCP failed")
continue
}
for _, p := range ps {
switch p.(type) {
case *rtcp.TransportLayerNack:
manager.logger.Info().Msg("got a nack")
manager.logger.Warn().Msg("got a nack")
}
}
}
}()
if err := session.SetPeer(&Peer{
id: id,
api: api,
@ -232,31 +243,30 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
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{
fba := []webrtc.RTCPFeedback{}
fbv := []webrtc.RTCPFeedback{
{"goog-remb", ""},
{"nack", ""},
{"nack", "pli"},
{"ccm", "fir"},
}
fba = []webrtc.RTCPFeedback{}
switch codecName {
case "VP8":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/VP8", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 96,}
codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/VP8", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fbv}, PayloadType: 96}
case "VP9":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/VP9", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fb}, PayloadType: 98,}
codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "video/VP9", ClockRate: 90000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fbv}, 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,}
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: fbv}, PayloadType: 102}
case "Opus":
codec = webrtc.RTPCodecParameters{ RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/opus", ClockRate: 48000, Channels: 2, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 111,}
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,}
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,}
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,}
codec = webrtc.RTPCodecParameters{RTPCodecCapability: webrtc.RTPCodecCapability{MimeType: "audio/PCMA", ClockRate: 8000, Channels: 0, SDPFmtpLine: "", RTCPFeedback: fba}, PayloadType: 8}
default:
return nil, codec, fmt.Errorf("unknown codec %s", codecName)
}