Squashed 'livekit-server/' content from commit 154b4d26

git-subtree-dir: livekit-server git-subtree-split: 154b4d26b769c68a03c096124094b97bf61a996f
2026-06-25 14:35:28 +09:00
commit 0da97ebd21
339 changed files with 114111 additions and 0 deletions
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+// Copyright 2023 LiveKit, Inc.
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+//     http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+
+package rtc
+
+import (
+	"github.com/pion/sdp/v3"
+	"github.com/pion/webrtc/v4"
+
+	"github.com/livekit/livekit-server/pkg/config"
+	"github.com/livekit/livekit-server/pkg/sfu/buffer"
+	dd "github.com/livekit/livekit-server/pkg/sfu/rtpextension/dependencydescriptor"
+	"github.com/livekit/mediatransportutil/pkg/rtcconfig"
+)
+
+const (
+	frameMarking        = "urn:ietf:params:rtp-hdrext:framemarking"
+	repairedRTPStreamID = "urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id"
+)
+
+type WebRTCConfig struct {
+	rtcconfig.WebRTCConfig
+
+	BufferFactory *buffer.Factory
+	Receiver      ReceiverConfig
+	Publisher     DirectionConfig
+	Subscriber    DirectionConfig
+}
+
+type ReceiverConfig struct {
+	PacketBufferSizeVideo int
+	PacketBufferSizeAudio int
+}
+
+type RTPHeaderExtensionConfig struct {
+	Audio []string
+	Video []string
+}
+
+type RTCPFeedbackConfig struct {
+	Audio []webrtc.RTCPFeedback
+	Video []webrtc.RTCPFeedback
+}
+
+type DirectionConfig struct {
+	RTPHeaderExtension RTPHeaderExtensionConfig
+	RTCPFeedback       RTCPFeedbackConfig
+}
+
+func NewWebRTCConfig(conf *config.Config) (*WebRTCConfig, error) {
+	rtcConf := conf.RTC
+
+	webRTCConfig, err := rtcconfig.NewWebRTCConfig(&rtcConf.RTCConfig, conf.Development)
+	if err != nil {
+		return nil, err
+	}
+
+	// we don't want to use active TCP on a server, clients should be dialing
+	webRTCConfig.SettingEngine.DisableActiveTCP(true)
+
+	if rtcConf.PacketBufferSize == 0 {
+		rtcConf.PacketBufferSize = 500
+	}
+	if rtcConf.PacketBufferSizeVideo == 0 {
+		rtcConf.PacketBufferSizeVideo = rtcConf.PacketBufferSize
+	}
+	if rtcConf.PacketBufferSizeAudio == 0 {
+		rtcConf.PacketBufferSizeAudio = rtcConf.PacketBufferSize
+	}
+
+	// publisher configuration
+	publisherConfig := DirectionConfig{
+		RTPHeaderExtension: RTPHeaderExtensionConfig{
+			Audio: []string{
+				sdp.SDESMidURI,
+				sdp.SDESRTPStreamIDURI,
+				sdp.AudioLevelURI,
+				//act.AbsCaptureTimeURI,
+			},
+			Video: []string{
+				sdp.SDESMidURI,
+				sdp.SDESRTPStreamIDURI,
+				sdp.TransportCCURI,
+				frameMarking,
+				dd.ExtensionURI,
+				repairedRTPStreamID,
+				//act.AbsCaptureTimeURI,
+			},
+		},
+		RTCPFeedback: RTCPFeedbackConfig{
+			Audio: []webrtc.RTCPFeedback{
+				{Type: webrtc.TypeRTCPFBNACK},
+			},
+			Video: []webrtc.RTCPFeedback{
+				{Type: webrtc.TypeRTCPFBTransportCC},
+				{Type: webrtc.TypeRTCPFBCCM, Parameter: "fir"},
+				{Type: webrtc.TypeRTCPFBNACK},
+				{Type: webrtc.TypeRTCPFBNACK, Parameter: "pli"},
+			},
+		},
+	}
+
+	return &WebRTCConfig{
+		WebRTCConfig: *webRTCConfig,
+		Receiver: ReceiverConfig{
+			PacketBufferSizeVideo: rtcConf.PacketBufferSizeVideo,
+			PacketBufferSizeAudio: rtcConf.PacketBufferSizeAudio,
+		},
+		Publisher:  publisherConfig,
+		Subscriber: getSubscriberConfig(rtcConf.CongestionControl.UseSendSideBWEInterceptor || rtcConf.CongestionControl.UseSendSideBWE),
+	}, nil
+}
+
+func (c *WebRTCConfig) UpdateCongestionControl(ccConf config.CongestionControlConfig) {
+	c.Subscriber = getSubscriberConfig(ccConf.UseSendSideBWEInterceptor || ccConf.UseSendSideBWE)
+}
+
+func (c *WebRTCConfig) SetBufferFactory(factory *buffer.Factory) {
+	c.BufferFactory = factory
+	c.SettingEngine.BufferFactory = factory.GetOrNew
+}
+
+func getSubscriberConfig(enableTWCC bool) DirectionConfig {
+	subscriberConfig := DirectionConfig{
+		RTPHeaderExtension: RTPHeaderExtensionConfig{
+			Video: []string{
+				dd.ExtensionURI,
+				//act.AbsCaptureTimeURI,
+			},
+			Audio: []string{
+				//act.AbsCaptureTimeURI,
+			},
+		},
+		RTCPFeedback: RTCPFeedbackConfig{
+			Audio: []webrtc.RTCPFeedback{
+				// always enable NACK for audio but disable it later for red enabled transceiver. https://github.com/pion/webrtc/pull/2972
+				{Type: webrtc.TypeRTCPFBNACK},
+			},
+			Video: []webrtc.RTCPFeedback{
+				{Type: webrtc.TypeRTCPFBCCM, Parameter: "fir"},
+				{Type: webrtc.TypeRTCPFBNACK},
+				{Type: webrtc.TypeRTCPFBNACK, Parameter: "pli"},
+			},
+		},
+	}
+	if enableTWCC {
+		subscriberConfig.RTPHeaderExtension.Video = append(subscriberConfig.RTPHeaderExtension.Video, sdp.TransportCCURI)
+		subscriberConfig.RTCPFeedback.Video = append(subscriberConfig.RTCPFeedback.Video, webrtc.RTCPFeedback{Type: webrtc.TypeRTCPFBTransportCC})
+	} else {
+		subscriberConfig.RTPHeaderExtension.Video = append(subscriberConfig.RTPHeaderExtension.Video, sdp.ABSSendTimeURI)
+		subscriberConfig.RTCPFeedback.Video = append(subscriberConfig.RTCPFeedback.Video, webrtc.RTCPFeedback{Type: webrtc.TypeRTCPFBGoogREMB})
+	}
+
+	return subscriberConfig
+}