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127 lines
3.4 KiB
Go
127 lines
3.4 KiB
Go
package rtc
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import (
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"github.com/pion/sdp/v3"
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"github.com/pion/webrtc/v3"
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"github.com/livekit/livekit-server/pkg/config"
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"github.com/livekit/livekit-server/pkg/sfu/buffer"
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dd "github.com/livekit/livekit-server/pkg/sfu/dependencydescriptor"
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"github.com/livekit/mediatransportutil/pkg/rtcconfig"
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)
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const (
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frameMarking = "urn:ietf:params:rtp-hdrext:framemarking"
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)
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type WebRTCConfig struct {
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rtcconfig.WebRTCConfig
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BufferFactory *buffer.Factory
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Receiver ReceiverConfig
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Publisher DirectionConfig
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Subscriber DirectionConfig
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}
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type ReceiverConfig struct {
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PacketBufferSize int
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}
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type RTPHeaderExtensionConfig struct {
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Audio []string
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Video []string
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}
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type RTCPFeedbackConfig struct {
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Audio []webrtc.RTCPFeedback
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Video []webrtc.RTCPFeedback
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}
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type DirectionConfig struct {
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RTPHeaderExtension RTPHeaderExtensionConfig
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RTCPFeedback RTCPFeedbackConfig
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StrictACKs bool
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}
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func NewWebRTCConfig(conf *config.Config) (*WebRTCConfig, error) {
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rtcConf := conf.RTC
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webRTCConfig, err := rtcconfig.NewWebRTCConfig(&rtcConf.RTCConfig, conf.Development)
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if err != nil {
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return nil, err
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}
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// we don't want to use active TCP on a server, clients should be dialing
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webRTCConfig.SettingEngine.DisableActiveTCP(true)
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if rtcConf.PacketBufferSize == 0 {
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rtcConf.PacketBufferSize = 500
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}
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// publisher configuration
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publisherConfig := DirectionConfig{
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StrictACKs: true, // publisher is dialed, and will always reply with ACK
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RTPHeaderExtension: RTPHeaderExtensionConfig{
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Audio: []string{
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sdp.SDESMidURI,
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sdp.SDESRTPStreamIDURI,
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sdp.AudioLevelURI,
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},
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Video: []string{
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sdp.SDESMidURI,
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sdp.SDESRTPStreamIDURI,
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sdp.TransportCCURI,
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frameMarking,
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dd.ExtensionUrl,
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},
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},
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RTCPFeedback: RTCPFeedbackConfig{
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Audio: []webrtc.RTCPFeedback{
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{Type: webrtc.TypeRTCPFBNACK},
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},
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Video: []webrtc.RTCPFeedback{
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{Type: webrtc.TypeRTCPFBTransportCC},
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{Type: webrtc.TypeRTCPFBCCM, Parameter: "fir"},
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{Type: webrtc.TypeRTCPFBNACK},
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{Type: webrtc.TypeRTCPFBNACK, Parameter: "pli"},
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},
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},
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}
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// subscriber configuration
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subscriberConfig := DirectionConfig{
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StrictACKs: conf.RTC.StrictACKs,
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RTPHeaderExtension: RTPHeaderExtensionConfig{
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Video: []string{dd.ExtensionUrl},
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},
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RTCPFeedback: RTCPFeedbackConfig{
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Video: []webrtc.RTCPFeedback{
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{Type: webrtc.TypeRTCPFBCCM, Parameter: "fir"},
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{Type: webrtc.TypeRTCPFBNACK},
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{Type: webrtc.TypeRTCPFBNACK, Parameter: "pli"},
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},
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},
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}
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if rtcConf.CongestionControl.UseSendSideBWE {
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subscriberConfig.RTPHeaderExtension.Video = append(subscriberConfig.RTPHeaderExtension.Video, sdp.TransportCCURI)
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subscriberConfig.RTCPFeedback.Video = append(subscriberConfig.RTCPFeedback.Video, webrtc.RTCPFeedback{Type: webrtc.TypeRTCPFBTransportCC})
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} else {
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subscriberConfig.RTPHeaderExtension.Video = append(subscriberConfig.RTPHeaderExtension.Video, sdp.ABSSendTimeURI)
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subscriberConfig.RTCPFeedback.Video = append(subscriberConfig.RTCPFeedback.Video, webrtc.RTCPFeedback{Type: webrtc.TypeRTCPFBGoogREMB})
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}
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return &WebRTCConfig{
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WebRTCConfig: *webRTCConfig,
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Receiver: ReceiverConfig{
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PacketBufferSize: rtcConf.PacketBufferSize,
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},
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Publisher: publisherConfig,
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Subscriber: subscriberConfig,
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}, nil
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}
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func (c *WebRTCConfig) SetBufferFactory(factory *buffer.Factory) {
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c.BufferFactory = factory
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c.SettingEngine.BufferFactory = factory.GetOrNew
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}
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