mirror of
https://github.com/langhuihui/monibuca.git
synced 2025-09-27 03:25:56 +08:00
533 lines
15 KiB
Go
533 lines
15 KiB
Go
package plugin_webrtc
|
||
|
||
import (
|
||
"embed"
|
||
"fmt"
|
||
"io"
|
||
"net"
|
||
"net/http"
|
||
"regexp"
|
||
"strconv"
|
||
"strings"
|
||
"time"
|
||
|
||
"github.com/pion/logging"
|
||
"github.com/pion/sdp/v3"
|
||
"m7s.live/v5/pkg/config"
|
||
|
||
"github.com/pion/interceptor"
|
||
. "github.com/pion/webrtc/v4"
|
||
"m7s.live/v5"
|
||
. "m7s.live/v5/pkg"
|
||
. "m7s.live/v5/plugin/webrtc/pkg"
|
||
)
|
||
|
||
var (
|
||
//go:embed web
|
||
web embed.FS
|
||
//go:embed default.yaml
|
||
defaultYaml m7s.DefaultYaml
|
||
reg_level = regexp.MustCompile("profile-level-id=(4.+f)")
|
||
_ = m7s.InstallPlugin[WebRTCPlugin](m7s.PluginMeta{
|
||
DefaultYaml: defaultYaml,
|
||
NewPuller: NewPuller,
|
||
NewPusher: NewPusher,
|
||
})
|
||
)
|
||
|
||
type WebRTCPlugin struct {
|
||
m7s.Plugin
|
||
ICEServers []ICEServer `desc:"ice服务器配置"`
|
||
Port string `default:"tcp:9000" desc:"监听端口"`
|
||
PLI time.Duration `default:"2s" desc:"发送PLI请求间隔"` // 视频流丢包后，发送PLI请求
|
||
EnableDC bool `default:"true" desc:"是否启用DataChannel"` // 在不支持编码格式的情况下是否启用DataChannel传输
|
||
MimeType []string `desc:"MimeType过滤列表，为空则不过滤"` // MimeType过滤列表，支持的格式如：video/H264, audio/opus
|
||
s SettingEngine
|
||
portMapping map[int]int // 内部端口到外部端口的映射
|
||
}
|
||
|
||
func (p *WebRTCPlugin) RegisterHandler() map[string]http.HandlerFunc {
|
||
return map[string]http.HandlerFunc{
|
||
"/test/{name}": p.testPage,
|
||
"/push/{streamPath...}": p.servePush,
|
||
"/play/{streamPath...}": p.servePlay,
|
||
}
|
||
}
|
||
|
||
func (p *WebRTCPlugin) NewLogger(scope string) logging.LeveledLogger {
|
||
return &LoggerTransform{Logger: p.Logger.With("scope", scope)}
|
||
}
|
||
|
||
// createMediaEngine 创建新的MediaEngine实例
|
||
func (p *WebRTCPlugin) createMediaEngine(ssd *sdp.SessionDescription) *MediaEngine {
|
||
m := &MediaEngine{}
|
||
|
||
// 如果没有提供SDP，则使用传统方式注册编解码器
|
||
if ssd == nil {
|
||
p.registerLegacyCodecs(m)
|
||
return m
|
||
}
|
||
|
||
// 从SDP中解析MediaDescription并注册编解码器
|
||
p.registerCodecsFromSDP(m, ssd)
|
||
|
||
return m
|
||
}
|
||
|
||
// registerLegacyCodecs 注册传统方式的编解码器（向后兼容）
|
||
func (p *WebRTCPlugin) registerLegacyCodecs(m *MediaEngine) {
|
||
// 注册基础编解码器
|
||
if defaultCodecs, err := GetDefaultCodecs(); err != nil {
|
||
p.Error("Failed to get default codecs", "error", err)
|
||
} else {
|
||
for _, codecWithType := range defaultCodecs {
|
||
// 检查MimeType过滤
|
||
if p.isMimeTypeAllowed(codecWithType.Codec.MimeType) {
|
||
if err := m.RegisterCodec(codecWithType.Codec, codecWithType.Type); err != nil {
|
||
p.Warn("Failed to register default codec", "error", err, "mimeType", codecWithType.Codec.MimeType)
|
||
} else {
|
||
p.Debug("Registered default codec", "mimeType", codecWithType.Codec.MimeType, "payloadType", codecWithType.Codec.PayloadType)
|
||
}
|
||
} else {
|
||
p.Debug("Default codec filtered", "mimeType", codecWithType.Codec.MimeType)
|
||
}
|
||
}
|
||
}
|
||
}
|
||
|
||
// registerCodecsFromSDP 从SDP的MediaDescription中注册编解码器
|
||
func (p *WebRTCPlugin) registerCodecsFromSDP(m *MediaEngine, ssd *sdp.SessionDescription) {
|
||
for _, md := range ssd.MediaDescriptions {
|
||
// 跳过非音视频媒体类型
|
||
if md.MediaName.Media != "audio" && md.MediaName.Media != "video" {
|
||
continue
|
||
}
|
||
|
||
// 解析每个format（编解码器）
|
||
for _, format := range md.MediaName.Formats {
|
||
codec := p.parseCodecFromSDP(md, format)
|
||
if codec == nil {
|
||
continue
|
||
}
|
||
|
||
// 检查MimeType过滤
|
||
if !p.isMimeTypeAllowed(codec.MimeType) {
|
||
p.Debug("MimeType filtered", "mimeType", codec.MimeType)
|
||
continue
|
||
}
|
||
|
||
// 确定编解码器类型
|
||
var codecType RTPCodecType
|
||
if md.MediaName.Media == "audio" {
|
||
codecType = RTPCodecTypeAudio
|
||
} else {
|
||
codecType = RTPCodecTypeVideo
|
||
}
|
||
|
||
// 注册编解码器
|
||
if err := m.RegisterCodec(*codec, codecType); err != nil {
|
||
p.Warn("Failed to register codec from SDP", "error", err, "mimeType", codec.MimeType)
|
||
} else {
|
||
p.Debug("Registered codec from SDP", "mimeType", codec.MimeType, "payloadType", codec.PayloadType)
|
||
}
|
||
}
|
||
}
|
||
}
|
||
|
||
// parseCodecFromSDP 从SDP的MediaDescription中解析单个编解码器
|
||
func (p *WebRTCPlugin) parseCodecFromSDP(md *sdp.MediaDescription, format string) *RTPCodecParameters {
|
||
var codecName string
|
||
var clockRate uint32
|
||
var channels uint16
|
||
var fmtpLine string
|
||
|
||
// 从rtpmap属性中解析编解码器名称、时钟频率和通道数
|
||
for _, attr := range md.Attributes {
|
||
if attr.Key == "rtpmap" && strings.HasPrefix(attr.Value, format+" ") {
|
||
// 格式：payloadType codecName/clockRate[/channels]
|
||
parts := strings.Split(attr.Value, " ")
|
||
if len(parts) >= 2 {
|
||
codecParts := strings.Split(parts[1], "/")
|
||
if len(codecParts) >= 2 {
|
||
codecName = strings.ToUpper(codecParts[0])
|
||
if rate, err := strconv.ParseUint(codecParts[1], 10, 32); err == nil {
|
||
clockRate = uint32(rate)
|
||
}
|
||
if len(codecParts) >= 3 {
|
||
if ch, err := strconv.ParseUint(codecParts[2], 10, 16); err == nil {
|
||
channels = uint16(ch)
|
||
}
|
||
}
|
||
}
|
||
}
|
||
break
|
||
}
|
||
}
|
||
|
||
// 从fmtp属性中解析格式参数
|
||
for _, attr := range md.Attributes {
|
||
if attr.Key == "fmtp" && strings.HasPrefix(attr.Value, format+" ") {
|
||
if spaceIdx := strings.Index(attr.Value, " "); spaceIdx >= 0 {
|
||
fmtpLine = attr.Value[spaceIdx+1:]
|
||
}
|
||
break
|
||
}
|
||
}
|
||
|
||
// 如果没有找到rtpmap，尝试静态payload类型
|
||
if codecName == "" {
|
||
codecName, clockRate, channels = p.getStaticPayloadInfo(format)
|
||
}
|
||
|
||
if codecName == "" {
|
||
return nil
|
||
}
|
||
|
||
// 构造MimeType
|
||
var mimeType string
|
||
if md.MediaName.Media == "audio" {
|
||
mimeType = "audio/" + codecName
|
||
} else {
|
||
mimeType = "video/" + codecName
|
||
}
|
||
|
||
// 解析PayloadType
|
||
payloadType, err := strconv.ParseUint(format, 10, 8)
|
||
if err != nil {
|
||
return nil
|
||
}
|
||
|
||
return &RTPCodecParameters{
|
||
RTPCodecCapability: RTPCodecCapability{
|
||
MimeType: mimeType,
|
||
ClockRate: clockRate,
|
||
Channels: channels,
|
||
SDPFmtpLine: fmtpLine,
|
||
RTCPFeedback: p.getDefaultRTCPFeedback(mimeType),
|
||
},
|
||
PayloadType: PayloadType(payloadType),
|
||
}
|
||
}
|
||
|
||
// getStaticPayloadInfo 获取静态payload类型的编解码器信息
|
||
func (p *WebRTCPlugin) getStaticPayloadInfo(format string) (string, uint32, uint16) {
|
||
switch format {
|
||
case "0":
|
||
return "PCMU", 8000, 1
|
||
case "8":
|
||
return "PCMA", 8000, 1
|
||
case "96":
|
||
return "H264", 90000, 0
|
||
case "97":
|
||
return "H264", 90000, 0
|
||
case "98":
|
||
return "H264", 90000, 0
|
||
case "111":
|
||
return "OPUS", 48000, 2
|
||
default:
|
||
return "", 0, 0
|
||
}
|
||
}
|
||
|
||
// getDefaultRTCPFeedback 获取默认的RTCP反馈
|
||
func (p *WebRTCPlugin) getDefaultRTCPFeedback(mimeType string) []RTCPFeedback {
|
||
if strings.HasPrefix(mimeType, "video/") {
|
||
return []RTCPFeedback{
|
||
{Type: "goog-remb", Parameter: ""},
|
||
{Type: "ccm", Parameter: "fir"},
|
||
{Type: "nack", Parameter: ""},
|
||
{Type: "nack", Parameter: "pli"},
|
||
{Type: "transport-cc", Parameter: ""},
|
||
}
|
||
}
|
||
return nil
|
||
}
|
||
|
||
// isMimeTypeAllowed 检查MimeType是否在允许列表中
|
||
func (p *WebRTCPlugin) isMimeTypeAllowed(mimeType string) bool {
|
||
// 如果过滤列表为空，则允许所有类型
|
||
if len(p.MimeType) == 0 {
|
||
return true
|
||
}
|
||
|
||
// 检查精确匹配
|
||
for _, filter := range p.MimeType {
|
||
if strings.EqualFold(filter, mimeType) {
|
||
return true
|
||
}
|
||
// 支持通配符匹配，如 "video/*" 或 "audio/*"
|
||
if strings.HasSuffix(filter, "/*") {
|
||
prefix := strings.TrimSuffix(filter, "/*")
|
||
if strings.HasPrefix(strings.ToLower(mimeType), strings.ToLower(prefix)+"/") {
|
||
return true
|
||
}
|
||
}
|
||
}
|
||
|
||
return false
|
||
}
|
||
|
||
// createAPI 创建新的API实例
|
||
func (p *WebRTCPlugin) createAPI(ssd *sdp.SessionDescription) (api *API, err error) {
|
||
m := p.createMediaEngine(ssd)
|
||
i := &interceptor.Registry{}
|
||
|
||
// 注册默认拦截器
|
||
if err := RegisterDefaultInterceptors(m, i); err != nil {
|
||
p.Error("register default interceptors error", "error", err)
|
||
return nil, err
|
||
}
|
||
|
||
// 创建API
|
||
api = NewAPI(WithMediaEngine(m), WithInterceptorRegistry(i), WithSettingEngine(p.s))
|
||
return
|
||
}
|
||
|
||
// initSettingEngine 初始化SettingEngine
|
||
func (p *WebRTCPlugin) initSettingEngine() error {
|
||
// 设置LoggerFactory
|
||
p.s.LoggerFactory = p
|
||
|
||
// 配置NAT 1:1 IP映射
|
||
if p.GetCommonConf().PublicIP != "" {
|
||
ips := []string{p.GetCommonConf().PublicIP}
|
||
if p.GetCommonConf().PublicIPv6 != "" {
|
||
ips = append(ips, p.GetCommonConf().PublicIPv6)
|
||
}
|
||
p.s.SetNAT1To1IPs(ips, ICECandidateTypeHost)
|
||
}
|
||
|
||
// 配置端口
|
||
if err := p.configurePort(); err != nil {
|
||
return err
|
||
}
|
||
|
||
return nil
|
||
}
|
||
|
||
// configurePort 配置端口设置
|
||
func (p *WebRTCPlugin) configurePort() error {
|
||
// 使用 ParsePort 而不是 ParsePort2 来获取端口映射信息
|
||
portInfo, err := ParsePort(p.Port)
|
||
if err != nil {
|
||
p.Error("webrtc port config error", "error", err, "port", p.Port)
|
||
return err
|
||
}
|
||
|
||
// 初始化端口映射
|
||
p.portMapping = make(map[int]int)
|
||
|
||
// 如果有端口映射，存储映射关系
|
||
if portInfo.HasMapping() {
|
||
if portInfo.IsRange() {
|
||
// 端口范围映射
|
||
for i := 0; i <= portInfo.Ports[1]-portInfo.Ports[0]; i++ {
|
||
internalPort := portInfo.Ports[0] + i
|
||
var externalPort int
|
||
if portInfo.IsRangeMapping() {
|
||
// 映射端口也是范围
|
||
externalPort = portInfo.Map[0] + i
|
||
} else {
|
||
// 映射端口是单个端口
|
||
externalPort = portInfo.Map[0]
|
||
}
|
||
p.portMapping[internalPort] = externalPort
|
||
}
|
||
} else {
|
||
// 单端口映射
|
||
p.portMapping[portInfo.Ports[0]] = portInfo.Map[0]
|
||
}
|
||
p.Info("Port mapping configured", "mapping", p.portMapping)
|
||
}
|
||
|
||
// 根据协议类型进行配置
|
||
if portInfo.IsTCP() {
|
||
if portInfo.IsRange() {
|
||
// TCP端口范围，这里可能需要特殊处理
|
||
p.Error("TCP port range not supported in current implementation")
|
||
return fmt.Errorf("TCP port range not supported")
|
||
} else {
|
||
// TCP单端口
|
||
tcpport := portInfo.Ports[0]
|
||
tcpl, err := net.ListenTCP("tcp", &net.TCPAddr{
|
||
IP: net.IP{0, 0, 0, 0},
|
||
Port: tcpport,
|
||
})
|
||
p.OnDispose(func() {
|
||
_ = tcpl.Close()
|
||
})
|
||
if err != nil {
|
||
p.Error("webrtc listener tcp", "error", err)
|
||
return err
|
||
}
|
||
p.SetDescription("tcp", fmt.Sprintf("%d", tcpport))
|
||
p.Info("webrtc start listen", "port", tcpport)
|
||
p.s.SetICETCPMux(NewICETCPMux(nil, tcpl, 4096))
|
||
p.s.SetNetworkTypes([]NetworkType{NetworkTypeTCP4, NetworkTypeTCP6})
|
||
p.s.DisableSRTPReplayProtection(true)
|
||
}
|
||
} else {
|
||
// UDP配置
|
||
if portInfo.IsRange() {
|
||
// UDP端口范围
|
||
p.s.SetEphemeralUDPPortRange(uint16(portInfo.Ports[0]), uint16(portInfo.Ports[1]))
|
||
p.SetDescription("udp", fmt.Sprintf("%d-%d", portInfo.Ports[0], portInfo.Ports[1]))
|
||
} else {
|
||
// UDP单端口
|
||
udpport := portInfo.Ports[0]
|
||
udpListener, err := net.ListenUDP("udp", &net.UDPAddr{
|
||
IP: net.IP{0, 0, 0, 0},
|
||
Port: udpport,
|
||
})
|
||
p.OnDispose(func() {
|
||
_ = udpListener.Close()
|
||
})
|
||
if err != nil {
|
||
p.Error("webrtc listener udp", "error", err)
|
||
return err
|
||
}
|
||
p.SetDescription("udp", fmt.Sprintf("%d", udpport))
|
||
p.Info("webrtc start listen", "port", udpport)
|
||
p.s.SetICEUDPMux(NewICEUDPMux(nil, udpListener))
|
||
p.s.SetNetworkTypes([]NetworkType{NetworkTypeUDP4, NetworkTypeUDP6})
|
||
}
|
||
}
|
||
|
||
return nil
|
||
}
|
||
|
||
func (p *WebRTCPlugin) CreatePC(sd SessionDescription, conf Configuration) (pc *PeerConnection, err error) {
|
||
var ssd *sdp.SessionDescription
|
||
ssd, err = sd.Unmarshal()
|
||
if err != nil {
|
||
return
|
||
}
|
||
var api *API
|
||
// 创建PeerConnection并设置高级配置
|
||
api, err = p.createAPI(ssd)
|
||
if err != nil {
|
||
return
|
||
}
|
||
pc, err = api.NewPeerConnection(conf)
|
||
if err != nil {
|
||
return
|
||
}
|
||
|
||
// 如果有端口映射配置，记录 ICE 候选者信息以供调试
|
||
if len(p.portMapping) > 0 {
|
||
pc.OnICECandidate(func(candidate *ICECandidate) {
|
||
if candidate != nil {
|
||
// 记录端口映射信息（用于调试和监控）
|
||
if mappedPort, exists := p.portMapping[int(candidate.Port)]; exists {
|
||
p.Debug("ICE candidate with port mapping detected",
|
||
"original_port", candidate.Port,
|
||
"mapped_port", mappedPort,
|
||
"candidate_address", candidate.Address,
|
||
"candidate_type", candidate.Typ)
|
||
candidate.Port = uint16(mappedPort) // 更新候选者端口为映射后的端口
|
||
} else {
|
||
p.Debug("ICE candidate generated",
|
||
"port", candidate.Port,
|
||
"address", candidate.Address,
|
||
"type", candidate.Typ)
|
||
}
|
||
}
|
||
})
|
||
}
|
||
|
||
err = pc.SetRemoteDescription(sd)
|
||
return
|
||
}
|
||
|
||
func (p *WebRTCPlugin) OnInit() (err error) {
|
||
if len(p.ICEServers) > 0 {
|
||
for i := range p.ICEServers {
|
||
b, _ := p.ICEServers[i].MarshalJSON()
|
||
p.ICEServers[i].UnmarshalJSON(b)
|
||
}
|
||
}
|
||
|
||
if err = p.initSettingEngine(); err != nil {
|
||
return err
|
||
}
|
||
|
||
_, port, _ := strings.Cut(p.GetCommonConf().HTTP.ListenAddr, ":")
|
||
if port == "80" {
|
||
p.PushAddr = append(p.PushAddr, "http://{hostName}/webrtc/push")
|
||
p.PlayAddr = append(p.PlayAddr, "http://{hostName}/webrtc/play")
|
||
} else if port != "" {
|
||
p.PushAddr = append(p.PushAddr, fmt.Sprintf("http://{hostName}:%s/webrtc/push", port))
|
||
p.PlayAddr = append(p.PlayAddr, fmt.Sprintf("http://{hostName}:%s/webrtc/play", port))
|
||
}
|
||
_, port, _ = strings.Cut(p.GetCommonConf().HTTP.ListenAddrTLS, ":")
|
||
if port == "443" {
|
||
p.PushAddr = append(p.PushAddr, "https://{hostName}/webrtc/push")
|
||
p.PlayAddr = append(p.PlayAddr, "https://{hostName}/webrtc/play")
|
||
} else if port != "" {
|
||
p.PushAddr = append(p.PushAddr, fmt.Sprintf("https://{hostName}:%s/webrtc/push", port))
|
||
p.PlayAddr = append(p.PlayAddr, fmt.Sprintf("https://{hostName}:%s/webrtc/play", port))
|
||
}
|
||
|
||
return
|
||
}
|
||
|
||
func (p *WebRTCPlugin) testPage(w http.ResponseWriter, r *http.Request) {
|
||
name := r.PathValue("name")
|
||
switch name {
|
||
case "publish", "screenshare":
|
||
name = "web/publish.html"
|
||
case "subscribe":
|
||
name = "web/subscribe.html"
|
||
case "push":
|
||
name = "web/push.html"
|
||
case "pull":
|
||
name = "web/pull.html"
|
||
case "batchv2":
|
||
name = "web/batchv2.html"
|
||
default:
|
||
name = "web/" + name
|
||
}
|
||
// Set appropriate MIME type based on file extension
|
||
if strings.HasSuffix(name, ".html") {
|
||
w.Header().Set("Content-Type", "text/html; charset=utf-8")
|
||
} else if strings.HasSuffix(name, ".js") {
|
||
w.Header().Set("Content-Type", "application/javascript")
|
||
// } else if strings.HasSuffix(name, ".css") {
|
||
// w.Header().Set("Content-Type", "text/css")
|
||
// } else if strings.HasSuffix(name, ".json") {
|
||
// w.Header().Set("Content-Type", "application/json")
|
||
// } else if strings.HasSuffix(name, ".png") {
|
||
// w.Header().Set("Content-Type", "image/png")
|
||
// } else if strings.HasSuffix(name, ".jpg") || strings.HasSuffix(name, ".jpeg") {
|
||
// w.Header().Set("Content-Type", "image/jpeg")
|
||
// } else if strings.HasSuffix(name, ".svg") {
|
||
// w.Header().Set("Content-Type", "image/svg+xml")
|
||
}
|
||
f, err := web.Open(name)
|
||
if err != nil {
|
||
http.Error(w, err.Error(), http.StatusNotFound)
|
||
return
|
||
}
|
||
io.Copy(w, f)
|
||
}
|
||
|
||
func (p *WebRTCPlugin) Pull(streamPath string, conf config.Pull, pubConf *config.Publish) {
|
||
if strings.HasPrefix(conf.URL, "https://rtc.live.cloudflare.com") {
|
||
cfClient := NewCFClient(DIRECTION_PULL)
|
||
api, err := p.createAPI(nil)
|
||
if err != nil {
|
||
p.Error("create API failed", "error", err)
|
||
return
|
||
}
|
||
cfClient.PeerConnection, err = api.NewPeerConnection(Configuration{
|
||
ICEServers: p.ICEServers,
|
||
BundlePolicy: BundlePolicyMaxBundle,
|
||
})
|
||
if err != nil {
|
||
p.Error("pull", "error", err)
|
||
return
|
||
}
|
||
cfClient.GetPullJob().Init(cfClient, &p.Plugin, streamPath, conf, pubConf)
|
||
}
|
||
}
|