fix logging for WebRTC.
This commit is contained in:
parent
8ef91be6ad
commit
29fc67aff9
@ -120,10 +120,15 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
|
|||||||
if err != nil {
|
if err != nil {
|
||||||
return "", manager.config.ICELite, manager.config.ICEServers, err
|
return "", manager.config.ICELite, manager.config.ICEServers, err
|
||||||
}
|
}
|
||||||
|
|
||||||
negotiated := true
|
negotiated := true
|
||||||
connection.CreateDataChannel("data", &webrtc.DataChannelInit{
|
_, err = connection.CreateDataChannel("data", &webrtc.DataChannelInit{
|
||||||
Negotiated: &negotiated,
|
Negotiated: &negotiated,
|
||||||
})
|
})
|
||||||
|
if err != nil {
|
||||||
|
return "", manager.config.ICELite, manager.config.ICEServers, err
|
||||||
|
}
|
||||||
|
|
||||||
connection.OnDataChannel(func(d *webrtc.DataChannel) {
|
connection.OnDataChannel(func(d *webrtc.DataChannel) {
|
||||||
d.OnMessage(func(msg webrtc.DataChannelMessage) {
|
d.OnMessage(func(msg webrtc.DataChannelMessage) {
|
||||||
if err = manager.handle(id, msg); err != nil {
|
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
|
// Set the handler for ICE connection state
|
||||||
// This will notify you when the peer has connected/disconnected
|
// This will notify you when the peer has connected/disconnected
|
||||||
connection.OnICEConnectionStateChange(func(connectionState webrtc.ICEConnectionState) {
|
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)
|
rtpSender, viderr := connection.AddTrack(manager.videoTrack)
|
||||||
@ -154,7 +161,7 @@ func (manager *WebRTCManager) CreatePeer(id string, session types.Session) (stri
|
|||||||
|
|
||||||
err = connection.SetLocalDescription(description)
|
err = connection.SetLocalDescription(description)
|
||||||
if err != nil {
|
if err != nil {
|
||||||
panic(err)
|
return "", manager.config.ICELite, manager.config.ICEServers, err
|
||||||
}
|
}
|
||||||
|
|
||||||
connection.OnConnectionStateChange(func(state webrtc.PeerConnectionState) {
|
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) {
|
connection.OnICECandidate(func(i *webrtc.ICECandidate) {
|
||||||
if i != nil {
|
if i == nil {
|
||||||
|
manager.logger.Info().Msg("sent all ICECandidates")
|
||||||
|
return
|
||||||
|
}
|
||||||
|
|
||||||
candidateString, err := json.Marshal(i.ToJSON())
|
candidateString, err := json.Marshal(i.ToJSON())
|
||||||
if err != nil {
|
if err != nil {
|
||||||
manager.logger.Info().Msg("error")
|
manager.logger.Warn().Err(err).Msg("converting ICECandidate to json failed")
|
||||||
return
|
return
|
||||||
}
|
}
|
||||||
|
|
||||||
if err = session.SignalCandidate(string(candidateString));err != nil {
|
if err := session.SignalCandidate(string(candidateString)); err != nil {
|
||||||
manager.logger.Info().Msg("err")
|
manager.logger.Warn().Err(err).Msg("sending SignalCandidate failed")
|
||||||
return
|
return
|
||||||
}
|
}
|
||||||
}
|
|
||||||
})
|
})
|
||||||
|
|
||||||
|
|
||||||
// Read incoming RTCP packets
|
// Read incoming RTCP packets
|
||||||
// Before these packets are retuned they are processed by interceptors. For things
|
// Before these packets are retuned they are processed by interceptors. For things
|
||||||
// like NACK this needs to be called.
|
// like NACK this needs to be called.
|
||||||
go func() {
|
go func() {
|
||||||
rtcpBuf := make([]byte, 1500)
|
rtcpBuf := make([]byte, 1500)
|
||||||
|
|
||||||
for {
|
for {
|
||||||
n, _, rtcpErr := rtpSender.Read(rtcpBuf)
|
n, _, err := rtpSender.Read(rtcpBuf)
|
||||||
if rtcpErr != nil {
|
if err != nil {
|
||||||
return
|
return
|
||||||
}
|
}
|
||||||
|
|
||||||
ps, err := rtcp.Unmarshal(rtcpBuf[:n])
|
ps, err := rtcp.Unmarshal(rtcpBuf[:n])
|
||||||
if err != nil {
|
if err != nil {
|
||||||
log.Printf("Unmarshal RTCP: %v", err)
|
manager.logger.Warn().Err(err).Msg("unmarshal RTCP failed")
|
||||||
continue
|
continue
|
||||||
}
|
}
|
||||||
|
|
||||||
for _, p := range ps {
|
for _, p := range ps {
|
||||||
switch p.(type) {
|
switch p.(type) {
|
||||||
case *rtcp.TransportLayerNack:
|
case *rtcp.TransportLayerNack:
|
||||||
manager.logger.Info().Msg("got a nack")
|
manager.logger.Warn().Msg("got a nack")
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
}()
|
}()
|
||||||
|
|
||||||
|
|
||||||
if err := session.SetPeer(&Peer{
|
if err := session.SetPeer(&Peer{
|
||||||
id: id,
|
id: id,
|
||||||
api: api,
|
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) {
|
func (m *WebRTCManager) createTrack(codecName string) (*webrtc.TrackLocalStaticSample, webrtc.RTPCodecParameters, error) {
|
||||||
var codec webrtc.RTPCodecParameters
|
var codec webrtc.RTPCodecParameters
|
||||||
var fb []webrtc.RTCPFeedback
|
|
||||||
var fba []webrtc.RTCPFeedback
|
fba := []webrtc.RTCPFeedback{}
|
||||||
fb = []webrtc.RTCPFeedback{
|
fbv := []webrtc.RTCPFeedback{
|
||||||
{"goog-remb", ""},
|
{"goog-remb", ""},
|
||||||
{"nack", ""},
|
{"nack", ""},
|
||||||
{"nack", "pli"},
|
{"nack", "pli"},
|
||||||
{"ccm", "fir"},
|
{"ccm", "fir"},
|
||||||
}
|
}
|
||||||
fba = []webrtc.RTCPFeedback{}
|
|
||||||
|
|
||||||
switch codecName {
|
switch codecName {
|
||||||
case "VP8":
|
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":
|
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":
|
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":
|
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":
|
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":
|
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":
|
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:
|
default:
|
||||||
return nil, codec, fmt.Errorf("unknown codec %s", codecName)
|
return nil, codec, fmt.Errorf("unknown codec %s", codecName)
|
||||||
}
|
}
|
||||||
|
Reference in New Issue
Block a user