微信小程序
This commit is contained in:
@@ -27,6 +27,7 @@ import (
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"xk-websocket-v2/internal/api"
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"xk-websocket-v2/internal/manager"
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"xk-websocket-v2/internal/mediaserver"
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"xk-websocket-v2/internal/middleware"
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"xk-websocket-v2/internal/model"
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"xk-websocket-v2/internal/service"
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@@ -229,6 +230,9 @@ func main() {
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// 步骤5: 启动TURN服务器（用于WebRTC音视频通话）
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go turnserver.Start()
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// 步骤5.5: 启动媒体服务器（WebRTC SFU + RTMP）
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go mediaserver.Start()
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// 步骤6: 创建Gin路由引擎
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r := gin.Default()
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@@ -391,6 +395,12 @@ func main() {
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// 朋友圈管理
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api.RegisterMomentRoutes(authGroup)
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// 通话管理（音视频）
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api.RegisterCallRoutes(authGroup)
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// 媒体服务器管理
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api.RegisterMediaRoutes(authGroup)
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}
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}
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@@ -57,3 +57,24 @@ turn:
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min_port: 49152
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# UDP 端口范围上限
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max_port: 65535
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# ==========================================
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# 媒体服务器配置 (WebRTC SFU + RTMP)
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# ==========================================
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media:
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# 是否启用媒体服务器
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enabled: true
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# 服务器公网 IP (默认复用 turn.public_ip)
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public_ip: ""
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# WebRTC SFU 端口
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webrtc_port: 8080
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# RTMP 服务端口
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rtmp_port: 1935
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# HTTP-FLV 端口
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http_flv_port: 7001
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# 单房间最大人数
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max_room_size: 9
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# 视频编解码器 (h264)
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video_codec: "h264"
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# 音频编解码器 (opus)
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audio_codec: "opus"
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15
go.mod
15
go.mod
@@ -8,7 +8,10 @@ require (
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github.com/golang-jwt/jwt/v5 v5.3.0
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github.com/gorilla/websocket v1.5.3
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github.com/panjf2000/ants/v2 v2.11.3
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github.com/pion/interceptor v0.1.29
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github.com/pion/rtp v1.8.7
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github.com/pion/turn/v2 v2.1.6
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github.com/pion/webrtc/v3 v3.3.6
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github.com/spf13/viper v1.21.0
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golang.org/x/crypto v0.45.0
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gorm.io/driver/mysql v1.6.0
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@@ -22,6 +25,7 @@ require (
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github.com/bytedance/sonic/loader v0.4.0 // indirect
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github.com/cespare/xxhash/v2 v2.3.0 // indirect
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github.com/cloudwego/base64x v0.1.6 // indirect
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github.com/davecgh/go-spew v1.1.1 // indirect
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github.com/dgryski/go-rendezvous v0.0.0-20200823014737-9f7001d12a5f // indirect
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github.com/fsnotify/fsnotify v1.9.0 // indirect
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github.com/gabriel-vasile/mimetype v1.4.11 // indirect
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@@ -33,6 +37,7 @@ require (
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github.com/go-viper/mapstructure/v2 v2.4.0 // indirect
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github.com/goccy/go-json v0.10.5 // indirect
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github.com/goccy/go-yaml v1.19.0 // indirect
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github.com/google/uuid v1.3.1 // indirect
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github.com/jinzhu/inflection v1.0.0 // indirect
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github.com/jinzhu/now v1.1.5 // indirect
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github.com/json-iterator/go v1.1.12 // indirect
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@@ -42,18 +47,27 @@ require (
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github.com/modern-go/concurrent v0.0.0-20180306012644-bacd9c7ef1dd // indirect
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github.com/modern-go/reflect2 v1.0.2 // indirect
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github.com/pelletier/go-toml/v2 v2.2.4 // indirect
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github.com/pion/datachannel v1.5.8 // indirect
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github.com/pion/dtls/v2 v2.2.12 // indirect
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github.com/pion/ice/v2 v2.3.38 // indirect
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github.com/pion/logging v0.2.4 // indirect
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github.com/pion/mdns v0.0.12 // indirect
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github.com/pion/randutil v0.1.0 // indirect
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github.com/pion/rtcp v1.2.14 // indirect
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github.com/pion/sctp v1.8.19 // indirect
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github.com/pion/sdp/v3 v3.0.9 // indirect
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github.com/pion/srtp/v2 v2.0.20 // indirect
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github.com/pion/stun v0.6.1 // indirect
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github.com/pion/transport/v2 v2.2.10 // indirect
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github.com/pion/transport/v3 v3.1.1 // indirect
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github.com/pmezard/go-difflib v1.0.0 // indirect
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github.com/quic-go/qpack v0.6.0 // indirect
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github.com/quic-go/quic-go v0.57.1 // indirect
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github.com/sagikazarmark/locafero v0.12.0 // indirect
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github.com/spf13/afero v1.15.0 // indirect
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github.com/spf13/cast v1.10.0 // indirect
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github.com/spf13/pflag v1.0.10 // indirect
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github.com/stretchr/testify v1.11.1 // indirect
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github.com/subosito/gotenv v1.6.0 // indirect
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github.com/twitchyliquid64/golang-asm v0.15.1 // indirect
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github.com/ugorji/go/codec v1.3.1 // indirect
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@@ -66,4 +80,5 @@ require (
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golang.org/x/sys v0.38.0 // indirect
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golang.org/x/text v0.31.0 // indirect
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google.golang.org/protobuf v1.36.10 // indirect
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gopkg.in/yaml.v3 v3.0.1 // indirect
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)
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32
go.sum
32
go.sum
@@ -48,6 +48,8 @@ github.com/golang-jwt/jwt/v5 v5.3.0/go.mod h1:fxCRLWMO43lRc8nhHWY6LGqRcf+1gQWArs
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github.com/google/go-cmp v0.7.0 h1:wk8382ETsv4JYUZwIsn6YpYiWiBsYLSJiTsyBybVuN8=
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github.com/google/go-cmp v0.7.0/go.mod h1:pXiqmnSA92OHEEa9HXL2W4E7lf9JzCmGVUdgjX3N/iU=
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github.com/google/gofuzz v1.0.0/go.mod h1:dBl0BpW6vV/+mYPU4Po3pmUjxk6FQPldtuIdl/M65Eg=
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github.com/google/uuid v1.3.1 h1:KjJaJ9iWZ3jOFZIf1Lqf4laDRCasjl0BCmnEGxkdLb4=
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github.com/google/uuid v1.3.1/go.mod h1:TIyPZe4MgqvfeYDBFedMoGGpEw/LqOeaOT+nhxU+yHo=
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github.com/gorilla/websocket v1.5.3 h1:saDtZ6Pbx/0u+bgYQ3q96pZgCzfhKXGPqt7kZ72aNNg=
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github.com/gorilla/websocket v1.5.3/go.mod h1:YR8l580nyteQvAITg2hZ9XVh4b55+EU/adAjf1fMHhE=
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github.com/jinzhu/inflection v1.0.0 h1:K317FqzuhWc8YvSVlFMCCUb36O/S9MCKRDI7QkRKD/E=
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@@ -58,8 +60,11 @@ github.com/json-iterator/go v1.1.12 h1:PV8peI4a0ysnczrg+LtxykD8LfKY9ML6u2jnxaEnr
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github.com/json-iterator/go v1.1.12/go.mod h1:e30LSqwooZae/UwlEbR2852Gd8hjQvJoHmT4TnhNGBo=
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github.com/klauspost/cpuid/v2 v2.3.0 h1:S4CRMLnYUhGeDFDqkGriYKdfoFlDnMtqTiI/sFzhA9Y=
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github.com/klauspost/cpuid/v2 v2.3.0/go.mod h1:hqwkgyIinND0mEev00jJYCxPNVRVXFQeu1XKlok6oO0=
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github.com/kr/pretty v0.1.0/go.mod h1:dAy3ld7l9f0ibDNOQOHHMYYIIbhfbHSm3C4ZsoJORNo=
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github.com/kr/pretty v0.3.1 h1:flRD4NNwYAUpkphVc1HcthR4KEIFJ65n8Mw5qdRn3LE=
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github.com/kr/pretty v0.3.1/go.mod h1:hoEshYVHaxMs3cyo3Yncou5ZscifuDolrwPKZanG3xk=
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github.com/kr/pty v1.1.1/go.mod h1:pFQYn66WHrOpPYNljwOMqo10TkYh1fy3cYio2l3bCsQ=
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github.com/kr/text v0.1.0/go.mod h1:4Jbv+DJW3UT/LiOwJeYQe1efqtUx/iVham/4vfdArNI=
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github.com/kr/text v0.2.0 h1:5Nx0Ya0ZqY2ygV366QzturHI13Jq95ApcVaJBhpS+AY=
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github.com/kr/text v0.2.0/go.mod h1:eLer722TekiGuMkidMxC/pM04lWEeraHUUmBw8l2grE=
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github.com/leodido/go-urn v1.4.0 h1:WT9HwE9SGECu3lg4d/dIA+jxlljEa1/ffXKmRjqdmIQ=
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@@ -81,24 +86,49 @@ github.com/panjf2000/ants/v2 v2.11.3 h1:AfI0ngBoXJmYOpDh9m516vjqoUu2sLrIVgppI9TZ
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github.com/panjf2000/ants/v2 v2.11.3/go.mod h1:8u92CYMUc6gyvTIw8Ru7Mt7+/ESnJahz5EVtqfrilek=
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github.com/pelletier/go-toml/v2 v2.2.4 h1:mye9XuhQ6gvn5h28+VilKrrPoQVanw5PMw/TB0t5Ec4=
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github.com/pelletier/go-toml/v2 v2.2.4/go.mod h1:2gIqNv+qfxSVS7cM2xJQKtLSTLUE9V8t9Stt+h56mCY=
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github.com/pion/datachannel v1.5.8 h1:ph1P1NsGkazkjrvyMfhRBUAWMxugJjq2HfQifaOoSNo=
|
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github.com/pion/datachannel v1.5.8/go.mod h1:PgmdpoaNBLX9HNzNClmdki4DYW5JtI7Yibu8QzbL3tI=
|
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github.com/pion/dtls/v2 v2.2.7/go.mod h1:8WiMkebSHFD0T+dIU+UeBaoV7kDhOW5oDCzZ7WZ/F9s=
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github.com/pion/dtls/v2 v2.2.12 h1:KP7H5/c1EiVAAKUmXyCzPiQe5+bCJrpOeKg/L05dunk=
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github.com/pion/dtls/v2 v2.2.12/go.mod h1:d9SYc9fch0CqK90mRk1dC7AkzzpwJj6u2GU3u+9pqFE=
|
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github.com/pion/ice/v2 v2.3.38 h1:DEpt13igPfvkE2+1Q+6e8mP30dtWnQD3CtMIKoRDRmA=
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github.com/pion/ice/v2 v2.3.38/go.mod h1:mBF7lnigdqgtB+YHkaY/Y6s6tsyRyo4u4rPGRuOjUBQ=
|
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github.com/pion/interceptor v0.1.29 h1:39fsnlP1U8gw2JzOFWdfCU82vHvhW9o0rZnZF56wF+M=
|
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github.com/pion/interceptor v0.1.29/go.mod h1:ri+LGNjRUc5xUNtDEPzfdkmSqISixVTBF/z/Zms/6T4=
|
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github.com/pion/logging v0.2.2/go.mod h1:k0/tDVsRCX2Mb2ZEmTqNa7CWsQPc+YYCB7Q+5pahoms=
|
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github.com/pion/logging v0.2.4 h1:tTew+7cmQ+Mc1pTBLKH2puKsOvhm32dROumOZ655zB8=
|
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github.com/pion/logging v0.2.4/go.mod h1:DffhXTKYdNZU+KtJ5pyQDjvOAh/GsNSyv1lbkFbe3so=
|
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github.com/pion/mdns v0.0.12 h1:CiMYlY+O0azojWDmxdNr7ADGrnZ+V6Ilfner+6mSVK8=
|
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github.com/pion/mdns v0.0.12/go.mod h1:VExJjv8to/6Wqm1FXK+Ii/Z9tsVk/F5sD/N70cnYFbk=
|
||||
github.com/pion/randutil v0.1.0 h1:CFG1UdESneORglEsnimhUjf33Rwjubwj6xfiOXBa3mA=
|
||||
github.com/pion/randutil v0.1.0/go.mod h1:XcJrSMMbbMRhASFVOlj/5hQial/Y8oH/HVo7TBZq+j8=
|
||||
github.com/pion/rtcp v1.2.12/go.mod h1:sn6qjxvnwyAkkPzPULIbVqSKI5Dv54Rv7VG0kNxh9L4=
|
||||
github.com/pion/rtcp v1.2.14 h1:KCkGV3vJ+4DAJmvP0vaQShsb0xkRfWkO540Gy102KyE=
|
||||
github.com/pion/rtcp v1.2.14/go.mod h1:sn6qjxvnwyAkkPzPULIbVqSKI5Dv54Rv7VG0kNxh9L4=
|
||||
github.com/pion/rtp v1.8.3/go.mod h1:pBGHaFt/yW7bf1jjWAoUjpSNoDnw98KTMg+jWWvziqU=
|
||||
github.com/pion/rtp v1.8.7 h1:qslKkG8qxvQ7hqaxkmL7Pl0XcUm+/Er7nMnu6Vq+ZxM=
|
||||
github.com/pion/rtp v1.8.7/go.mod h1:pBGHaFt/yW7bf1jjWAoUjpSNoDnw98KTMg+jWWvziqU=
|
||||
github.com/pion/sctp v1.8.19 h1:2CYuw+SQ5vkQ9t0HdOPccsCz1GQMDuVy5PglLgKVBW8=
|
||||
github.com/pion/sctp v1.8.19/go.mod h1:P6PbDVA++OJMrVNg2AL3XtYHV4uD6dvfyOovCgMs0PE=
|
||||
github.com/pion/sdp/v3 v3.0.9 h1:pX++dCHoHUwq43kuwf3PyJfHlwIj4hXA7Vrifiq0IJY=
|
||||
github.com/pion/sdp/v3 v3.0.9/go.mod h1:B5xmvENq5IXJimIO4zfp6LAe1fD9N+kFv+V/1lOdz8M=
|
||||
github.com/pion/srtp/v2 v2.0.20 h1:HNNny4s+OUmG280ETrCdgFndp4ufx3/uy85EawYEhTk=
|
||||
github.com/pion/srtp/v2 v2.0.20/go.mod h1:0KJQjA99A6/a0DOVTu1PhDSw0CXF2jTkqOoMg3ODqdA=
|
||||
github.com/pion/stun v0.6.1 h1:8lp6YejULeHBF8NmV8e2787BogQhduZugh5PdhDyyN4=
|
||||
github.com/pion/stun v0.6.1/go.mod h1:/hO7APkX4hZKu/D0f2lHzNyvdkTGtIy3NDmLR7kSz/8=
|
||||
github.com/pion/transport/v2 v2.2.1/go.mod h1:cXXWavvCnFF6McHTft3DWS9iic2Mftcz1Aq29pGcU5g=
|
||||
github.com/pion/transport/v2 v2.2.3/go.mod h1:q2U/tf9FEfnSBGSW6w5Qp5PFWRLRj3NjLhCCgpRK4p0=
|
||||
github.com/pion/transport/v2 v2.2.4/go.mod h1:q2U/tf9FEfnSBGSW6w5Qp5PFWRLRj3NjLhCCgpRK4p0=
|
||||
github.com/pion/transport/v2 v2.2.10 h1:ucLBLE8nuxiHfvkFKnkDQRYWYfp8ejf4YBOPfaQpw6Q=
|
||||
github.com/pion/transport/v2 v2.2.10/go.mod h1:sq1kSLWs+cHW9E+2fJP95QudkzbK7wscs8yYgQToO5E=
|
||||
github.com/pion/transport/v3 v3.0.1/go.mod h1:UY7kiITrlMv7/IKgd5eTUcaahZx5oUN3l9SzK5f5xE0=
|
||||
github.com/pion/transport/v3 v3.1.1 h1:Tr684+fnnKlhPceU+ICdrw6KKkTms+5qHMgw6bIkYOM=
|
||||
github.com/pion/transport/v3 v3.1.1/go.mod h1:+c2eewC5WJQHiAA46fkMMzoYZSuGzA/7E2FPrOYHctQ=
|
||||
github.com/pion/turn/v2 v2.1.3/go.mod h1:huEpByKKHix2/b9kmTAM3YoX6MKP+/D//0ClgUYR2fY=
|
||||
github.com/pion/turn/v2 v2.1.6 h1:Xr2niVsiPTB0FPtt+yAWKFUkU1eotQbGgpTIld4x1Gc=
|
||||
github.com/pion/turn/v2 v2.1.6/go.mod h1:huEpByKKHix2/b9kmTAM3YoX6MKP+/D//0ClgUYR2fY=
|
||||
github.com/pion/webrtc/v3 v3.3.6 h1:7XAh4RPtlY1Vul6/GmZrv7z+NnxKA6If0KStXBI2ZLE=
|
||||
github.com/pion/webrtc/v3 v3.3.6/go.mod h1:zyN7th4mZpV27eXybfR/cnUf3J2DRy8zw/mdjD9JTNM=
|
||||
github.com/pmezard/go-difflib v1.0.0 h1:4DBwDE0NGyQoBHbLQYPwSUPoCMWR5BEzIk/f1lZbAQM=
|
||||
github.com/pmezard/go-difflib v1.0.0/go.mod h1:iKH77koFhYxTK1pcRnkKkqfTogsbg7gZNVY4sRDYZ/4=
|
||||
github.com/quic-go/qpack v0.6.0 h1:g7W+BMYynC1LbYLSqRt8PBg5Tgwxn214ZZR34VIOjz8=
|
||||
@@ -126,6 +156,7 @@ github.com/stretchr/testify v1.7.1/go.mod h1:6Fq8oRcR53rry900zMqJjRRixrwX3KX962/
|
||||
github.com/stretchr/testify v1.8.0/go.mod h1:yNjHg4UonilssWZ8iaSj1OCr/vHnekPRkoO+kdMU+MU=
|
||||
github.com/stretchr/testify v1.8.3/go.mod h1:sz/lmYIOXD/1dqDmKjjqLyZ2RngseejIcXlSw2iwfAo=
|
||||
github.com/stretchr/testify v1.8.4/go.mod h1:sz/lmYIOXD/1dqDmKjjqLyZ2RngseejIcXlSw2iwfAo=
|
||||
github.com/stretchr/testify v1.9.0/go.mod h1:r2ic/lqez/lEtzL7wO/rwa5dbSLXVDPFyf8C91i36aY=
|
||||
github.com/stretchr/testify v1.10.0/go.mod h1:r2ic/lqez/lEtzL7wO/rwa5dbSLXVDPFyf8C91i36aY=
|
||||
github.com/stretchr/testify v1.11.1 h1:7s2iGBzp5EwR7/aIZr8ao5+dra3wiQyKjjFuvgVKu7U=
|
||||
github.com/stretchr/testify v1.11.1/go.mod h1:wZwfW3scLgRK+23gO65QZefKpKQRnfz6sD981Nm4B6U=
|
||||
@@ -209,6 +240,7 @@ golang.org/x/xerrors v0.0.0-20190717185122-a985d3407aa7/go.mod h1:I/5z698sn9Ka8T
|
||||
google.golang.org/protobuf v1.36.10 h1:AYd7cD/uASjIL6Q9LiTjz8JLcrh/88q5UObnmY3aOOE=
|
||||
google.golang.org/protobuf v1.36.10/go.mod h1:HTf+CrKn2C3g5S8VImy6tdcUvCska2kB7j23XfzDpco=
|
||||
gopkg.in/check.v1 v0.0.0-20161208181325-20d25e280405/go.mod h1:Co6ibVJAznAaIkqp8huTwlJQCZ016jof/cbN4VW5Yz0=
|
||||
gopkg.in/check.v1 v1.0.0-20190902080502-41f04d3bba15/go.mod h1:Co6ibVJAznAaIkqp8huTwlJQCZ016jof/cbN4VW5Yz0=
|
||||
gopkg.in/check.v1 v1.0.0-20201130134442-10cb98267c6c h1:Hei/4ADfdWqJk1ZMxUNpqntNwaWcugrBjAiHlqqRiVk=
|
||||
gopkg.in/check.v1 v1.0.0-20201130134442-10cb98267c6c/go.mod h1:JHkPIbrfpd72SG/EVd6muEfDQjcINNoR0C8j2r3qZ4Q=
|
||||
gopkg.in/tomb.v1 v1.0.0-20141024135613-dd632973f1e7 h1:uRGJdciOHaEIrze2W8Q3AKkepLTh2hOroT7a+7czfdQ=
|
||||
|
||||
419
internal/api/call_handler.go
Normal file
419
internal/api/call_handler.go
Normal file
@@ -0,0 +1,419 @@
|
||||
/**
|
||||
* package api
|
||||
*
|
||||
* 通话相关 API 处理器
|
||||
* 功能：
|
||||
* 1. 创建/加入/离开通话房间
|
||||
* 2. 处理 WebRTC 信令（offer/answer/ice）
|
||||
* 3. 生成 RTMP 推拉流地址（小程序）
|
||||
*/
|
||||
package api
|
||||
|
||||
import (
|
||||
"fmt"
|
||||
|
||||
"xk-websocket-v2/internal/mediaserver"
|
||||
"xk-websocket-v2/internal/utils"
|
||||
|
||||
"github.com/gin-gonic/gin"
|
||||
"github.com/spf13/viper"
|
||||
)
|
||||
|
||||
// ========== 请求/响应结构体 ==========
|
||||
|
||||
// CreateCallRoomRequest 创建通话房间请求
|
||||
type CreateCallRoomRequest struct {
|
||||
RoomID string `json:"room_id" binding:"required"`
|
||||
CallType string `json:"call_type" binding:"required,oneof=audio video"` // audio/video
|
||||
IsGroupCall bool `json:"is_group_call"`
|
||||
}
|
||||
|
||||
// JoinCallRoomRequest 加入通话房间请求
|
||||
type JoinCallRoomRequest struct {
|
||||
RoomID string `json:"room_id" binding:"required"`
|
||||
UserID string `json:"user_id" binding:"required"`
|
||||
Platform string `json:"platform" binding:"required,oneof=h5 app miniprogram"` // h5/app/miniprogram
|
||||
}
|
||||
|
||||
// LeaveCallRoomRequest 离开通话房间请求
|
||||
type LeaveCallRoomRequest struct {
|
||||
RoomID string `json:"room_id" binding:"required"`
|
||||
UserID string `json:"user_id" binding:"required"`
|
||||
}
|
||||
|
||||
// WebRTCOfferRequest WebRTC Offer 请求
|
||||
type WebRTCOfferRequest struct {
|
||||
RoomID string `json:"room_id" binding:"required"`
|
||||
UserID string `json:"user_id" binding:"required"`
|
||||
SDP string `json:"sdp" binding:"required"`
|
||||
}
|
||||
|
||||
// WebRTCICERequest WebRTC ICE 候选请求
|
||||
type WebRTCICERequest struct {
|
||||
RoomID string `json:"room_id" binding:"required"`
|
||||
UserID string `json:"user_id" binding:"required"`
|
||||
Candidate string `json:"candidate" binding:"required"`
|
||||
}
|
||||
|
||||
// JoinCallRoomResponse 加入通话房间响应
|
||||
type JoinCallRoomResponse struct {
|
||||
RoomID string `json:"room_id"`
|
||||
Platform string `json:"platform"`
|
||||
ICEServers []ICEServerConfig `json:"ice_servers,omitempty"` // H5/App 用
|
||||
PushURL string `json:"push_url,omitempty"` // 小程序用
|
||||
PullURLs []PullURLInfo `json:"pull_urls,omitempty"` // 小程序用
|
||||
Participants []ParticipantInfo `json:"participants"`
|
||||
}
|
||||
|
||||
// ICEServerConfig ICE 服务器配置
|
||||
type ICEServerConfig struct {
|
||||
URLs []string `json:"urls"`
|
||||
Username string `json:"username,omitempty"`
|
||||
Credential string `json:"credential,omitempty"`
|
||||
}
|
||||
|
||||
// PullURLInfo 拉流地址信息
|
||||
type PullURLInfo struct {
|
||||
UserID string `json:"user_id"`
|
||||
URL string `json:"url"`
|
||||
FLVURL string `json:"flv_url,omitempty"`
|
||||
}
|
||||
|
||||
// ParticipantInfo 参与者信息
|
||||
type ParticipantInfo struct {
|
||||
UserID string `json:"user_id"`
|
||||
Platform string `json:"platform"`
|
||||
HasAudio bool `json:"has_audio"`
|
||||
HasVideo bool `json:"has_video"`
|
||||
}
|
||||
|
||||
// RoomInfoResponse 房间信息响应
|
||||
type RoomInfoResponse struct {
|
||||
RoomID string `json:"room_id"`
|
||||
CallType string `json:"call_type"`
|
||||
IsGroupCall bool `json:"is_group_call"`
|
||||
ParticipantCount int `json:"participant_count"`
|
||||
Participants []ParticipantInfo `json:"participants"`
|
||||
}
|
||||
|
||||
// ========== API 处理函数 ==========
|
||||
|
||||
// CreateCallRoomHandler 创建通话房间
|
||||
// POST /api/call/room
|
||||
func CreateCallRoomHandler(c *gin.Context) {
|
||||
var req CreateCallRoomRequest
|
||||
if err := c.ShouldBindJSON(&req); err != nil {
|
||||
utils.BadRequest(c, "参数错误: "+err.Error())
|
||||
return
|
||||
}
|
||||
|
||||
ms := mediaserver.GetServer()
|
||||
if ms.GetConfig() == nil || !ms.GetConfig().Enabled {
|
||||
utils.Error(c, 503, "媒体服务未启用")
|
||||
return
|
||||
}
|
||||
|
||||
// 创建房间
|
||||
room := ms.GetOrCreateRoom(req.RoomID)
|
||||
room.SetCallType(req.CallType, req.IsGroupCall)
|
||||
|
||||
utils.SuccessWithData(c, gin.H{
|
||||
"room_id": req.RoomID,
|
||||
"call_type": req.CallType,
|
||||
"is_group": req.IsGroupCall,
|
||||
"created": true,
|
||||
}, "创建成功")
|
||||
}
|
||||
|
||||
// JoinCallRoomHandler 加入通话房间
|
||||
// POST /api/call/join
|
||||
func JoinCallRoomHandler(c *gin.Context) {
|
||||
var req JoinCallRoomRequest
|
||||
if err := c.ShouldBindJSON(&req); err != nil {
|
||||
utils.BadRequest(c, "参数错误: "+err.Error())
|
||||
return
|
||||
}
|
||||
|
||||
ms := mediaserver.GetServer()
|
||||
if ms.GetConfig() == nil || !ms.GetConfig().Enabled {
|
||||
utils.Error(c, 503, "媒体服务未启用")
|
||||
return
|
||||
}
|
||||
|
||||
room := ms.GetRoom(req.RoomID)
|
||||
if room == nil {
|
||||
// 自动创建房间
|
||||
room = ms.GetOrCreateRoom(req.RoomID)
|
||||
}
|
||||
|
||||
response := JoinCallRoomResponse{
|
||||
RoomID: req.RoomID,
|
||||
Platform: req.Platform,
|
||||
}
|
||||
|
||||
switch req.Platform {
|
||||
case "h5", "app":
|
||||
// H5/App 使用 WebRTC
|
||||
pType := mediaserver.ParticipantTypeWebRTC
|
||||
_, err := room.AddParticipant(req.UserID, pType)
|
||||
if err != nil {
|
||||
utils.BadRequest(c, err.Error())
|
||||
return
|
||||
}
|
||||
|
||||
// 返回 ICE 服务器配置
|
||||
response.ICEServers = getICEServers(req.UserID)
|
||||
|
||||
case "miniprogram":
|
||||
// 小程序使用 RTMP
|
||||
pType := mediaserver.ParticipantTypeRTMP
|
||||
participant, err := room.AddParticipant(req.UserID, pType)
|
||||
if err != nil {
|
||||
utils.BadRequest(c, err.Error())
|
||||
return
|
||||
}
|
||||
|
||||
// 生成推拉流地址
|
||||
rtmpServer := ms.GetRTMP()
|
||||
if rtmpServer == nil {
|
||||
utils.Error(c, 503, "RTMP服务未启用")
|
||||
return
|
||||
}
|
||||
|
||||
stream, err := rtmpServer.GenerateStreamURLs(req.RoomID, req.UserID)
|
||||
if err != nil {
|
||||
utils.InternalError(c, "生成推流地址失败")
|
||||
return
|
||||
}
|
||||
|
||||
// 更新参与者的流地址
|
||||
room.SetParticipantRTMPURLs(req.UserID, stream.PushURL, stream.PullURL, stream.ID)
|
||||
participant.PushURL = stream.PushURL
|
||||
participant.PullURL = stream.PullURL
|
||||
participant.StreamID = stream.ID
|
||||
|
||||
response.PushURL = stream.PushURL
|
||||
|
||||
// 获取房间内其他用户的拉流地址
|
||||
pullURLs := make([]PullURLInfo, 0)
|
||||
for _, p := range room.GetOtherParticipants(req.UserID) {
|
||||
if p.PullURL != "" {
|
||||
pullURLs = append(pullURLs, PullURLInfo{
|
||||
UserID: p.UserID,
|
||||
URL: p.PullURL,
|
||||
})
|
||||
}
|
||||
}
|
||||
response.PullURLs = pullURLs
|
||||
}
|
||||
|
||||
// 获取参与者列表
|
||||
participants := make([]ParticipantInfo, 0)
|
||||
for _, p := range room.GetAllParticipants() {
|
||||
platform := "h5"
|
||||
if p.Type == mediaserver.ParticipantTypeRTMP {
|
||||
platform = "miniprogram"
|
||||
}
|
||||
participants = append(participants, ParticipantInfo{
|
||||
UserID: p.UserID,
|
||||
Platform: platform,
|
||||
HasAudio: p.HasAudio,
|
||||
HasVideo: p.HasVideo,
|
||||
})
|
||||
}
|
||||
response.Participants = participants
|
||||
|
||||
utils.SuccessWithData(c, response, "加入成功")
|
||||
}
|
||||
|
||||
// LeaveCallRoomHandler 离开通话房间
|
||||
// POST /api/call/leave
|
||||
func LeaveCallRoomHandler(c *gin.Context) {
|
||||
var req LeaveCallRoomRequest
|
||||
if err := c.ShouldBindJSON(&req); err != nil {
|
||||
utils.BadRequest(c, "参数错误: "+err.Error())
|
||||
return
|
||||
}
|
||||
|
||||
ms := mediaserver.GetServer()
|
||||
room := ms.GetRoom(req.RoomID)
|
||||
if room == nil {
|
||||
utils.SuccessWithData(c, gin.H{"left": true}, "已离开")
|
||||
return
|
||||
}
|
||||
|
||||
// 获取参与者信息
|
||||
participant := room.GetParticipant(req.UserID)
|
||||
if participant != nil && participant.Type == mediaserver.ParticipantTypeRTMP {
|
||||
// 清理 RTMP 流
|
||||
rtmpServer := ms.GetRTMP()
|
||||
if rtmpServer != nil && participant.StreamID != "" {
|
||||
rtmpServer.RemoveStream(participant.StreamID)
|
||||
}
|
||||
}
|
||||
|
||||
// 移除参与者
|
||||
room.RemoveParticipant(req.UserID)
|
||||
|
||||
// 如果房间空了，移除房间
|
||||
if room.IsEmpty() {
|
||||
ms.RemoveRoom(req.RoomID)
|
||||
}
|
||||
|
||||
utils.SuccessWithData(c, gin.H{"left": true}, "已离开")
|
||||
}
|
||||
|
||||
// GetCallRoomHandler 获取通话房间信息
|
||||
// GET /api/call/room/:room_id
|
||||
func GetCallRoomHandler(c *gin.Context) {
|
||||
roomID := c.Param("room_id")
|
||||
if roomID == "" {
|
||||
utils.BadRequest(c, "房间ID不能为空")
|
||||
return
|
||||
}
|
||||
|
||||
ms := mediaserver.GetServer()
|
||||
room := ms.GetRoom(roomID)
|
||||
if room == nil {
|
||||
utils.NotFound(c, "房间不存在")
|
||||
return
|
||||
}
|
||||
|
||||
info := room.GetInfo()
|
||||
|
||||
participants := make([]ParticipantInfo, 0)
|
||||
for _, p := range info.Participants {
|
||||
platform := "h5"
|
||||
if p.Type == mediaserver.ParticipantTypeRTMP {
|
||||
platform = "miniprogram"
|
||||
}
|
||||
participants = append(participants, ParticipantInfo{
|
||||
UserID: p.UserID,
|
||||
Platform: platform,
|
||||
HasAudio: p.HasAudio,
|
||||
HasVideo: p.HasVideo,
|
||||
})
|
||||
}
|
||||
|
||||
utils.SuccessWithData(c, RoomInfoResponse{
|
||||
RoomID: info.ID,
|
||||
CallType: info.CallType,
|
||||
IsGroupCall: info.IsGroupCall,
|
||||
ParticipantCount: info.ParticipantCount,
|
||||
Participants: participants,
|
||||
}, "获取成功")
|
||||
}
|
||||
|
||||
// WebRTCOfferHandler 处理 WebRTC Offer
|
||||
// POST /api/call/offer
|
||||
func WebRTCOfferHandler(c *gin.Context) {
|
||||
var req WebRTCOfferRequest
|
||||
if err := c.ShouldBindJSON(&req); err != nil {
|
||||
utils.BadRequest(c, "参数错误: "+err.Error())
|
||||
return
|
||||
}
|
||||
|
||||
ms := mediaserver.GetServer()
|
||||
sfu := ms.GetSFU()
|
||||
if sfu == nil {
|
||||
utils.Error(c, 503, "SFU服务未启用")
|
||||
return
|
||||
}
|
||||
|
||||
// 处理 Offer 并返回 Answer
|
||||
answerSDP, err := sfu.HandleOffer(req.RoomID, req.UserID, req.SDP)
|
||||
if err != nil {
|
||||
utils.InternalError(c, "处理Offer失败: "+err.Error())
|
||||
return
|
||||
}
|
||||
|
||||
utils.SuccessWithData(c, gin.H{
|
||||
"sdp": answerSDP,
|
||||
}, "处理成功")
|
||||
}
|
||||
|
||||
// WebRTCICEHandler 处理 WebRTC ICE 候选
|
||||
// POST /api/call/ice
|
||||
func WebRTCICEHandler(c *gin.Context) {
|
||||
var req WebRTCICERequest
|
||||
if err := c.ShouldBindJSON(&req); err != nil {
|
||||
utils.BadRequest(c, "参数错误: "+err.Error())
|
||||
return
|
||||
}
|
||||
|
||||
ms := mediaserver.GetServer()
|
||||
sfu := ms.GetSFU()
|
||||
if sfu == nil {
|
||||
utils.Error(c, 503, "SFU服务未启用")
|
||||
return
|
||||
}
|
||||
|
||||
// 添加 ICE 候选
|
||||
if err := sfu.HandleICECandidate(req.RoomID, req.UserID, req.Candidate); err != nil {
|
||||
utils.InternalError(c, "处理ICE失败: "+err.Error())
|
||||
return
|
||||
}
|
||||
|
||||
utils.SuccessWithData(c, gin.H{"added": true}, "添加成功")
|
||||
}
|
||||
|
||||
// GetICEServersHandler 获取 ICE 服务器配置
|
||||
// GET /api/call/ice-servers
|
||||
func GetICEServersHandler(c *gin.Context) {
|
||||
userID := c.Query("user_id")
|
||||
if userID == "" {
|
||||
userID = "anonymous"
|
||||
}
|
||||
|
||||
servers := getICEServers(userID)
|
||||
utils.SuccessWithData(c, gin.H{
|
||||
"ice_servers": servers,
|
||||
}, "获取成功")
|
||||
}
|
||||
|
||||
// getICEServers 获取 ICE 服务器配置
|
||||
func getICEServers(userID string) []ICEServerConfig {
|
||||
servers := []ICEServerConfig{}
|
||||
|
||||
turnPublicIP := viper.GetString("turn.public_ip")
|
||||
turnPort := viper.GetInt("turn.listen_port")
|
||||
|
||||
if turnPublicIP != "" && turnPort > 0 {
|
||||
portStr := fmt.Sprintf("%d", turnPort)
|
||||
|
||||
// STUN 服务器
|
||||
servers = append(servers, ICEServerConfig{
|
||||
URLs: []string{
|
||||
"stun:" + turnPublicIP + ":" + portStr,
|
||||
},
|
||||
})
|
||||
|
||||
// TURN 服务器（如果启用）
|
||||
if viper.GetBool("turn.enabled") {
|
||||
username, credential := mediaserver.GetTURNCredentials(userID)
|
||||
servers = append(servers, ICEServerConfig{
|
||||
URLs: []string{
|
||||
"turn:" + turnPublicIP + ":" + portStr,
|
||||
},
|
||||
Username: username,
|
||||
Credential: credential,
|
||||
})
|
||||
}
|
||||
}
|
||||
|
||||
return servers
|
||||
}
|
||||
|
||||
// RegisterCallRoutes 注册通话相关路由
|
||||
func RegisterCallRoutes(router *gin.RouterGroup) {
|
||||
callGroup := router.Group("/call")
|
||||
{
|
||||
callGroup.POST("/room", CreateCallRoomHandler)
|
||||
callGroup.GET("/room/:room_id", GetCallRoomHandler)
|
||||
callGroup.POST("/join", JoinCallRoomHandler)
|
||||
callGroup.POST("/leave", LeaveCallRoomHandler)
|
||||
callGroup.POST("/offer", WebRTCOfferHandler)
|
||||
callGroup.POST("/ice", WebRTCICEHandler)
|
||||
callGroup.GET("/ice-servers", GetICEServersHandler)
|
||||
}
|
||||
}
|
||||
183
internal/api/media_handler.go
Normal file
183
internal/api/media_handler.go
Normal file
@@ -0,0 +1,183 @@
|
||||
/**
|
||||
* package api
|
||||
*
|
||||
* 媒体服务监控 API 处理器
|
||||
* 功能：
|
||||
* 1. 查看媒体服务器状态
|
||||
* 2. 查看活跃房间和流
|
||||
* 3. 管理桥接器
|
||||
*/
|
||||
package api
|
||||
|
||||
import (
|
||||
"xk-websocket-v2/internal/mediaserver"
|
||||
"xk-websocket-v2/internal/utils"
|
||||
|
||||
"github.com/gin-gonic/gin"
|
||||
)
|
||||
|
||||
// MediaServerStatusResponse 媒体服务器状态响应
|
||||
type MediaServerStatusResponse struct {
|
||||
Enabled bool `json:"enabled"`
|
||||
Config map[string]interface{} `json:"config"`
|
||||
RoomCount int `json:"room_count"`
|
||||
SFUTrackCount int `json:"sfu_track_count"`
|
||||
RTMPStreamCount int `json:"rtmp_stream_count"`
|
||||
}
|
||||
|
||||
// GetMediaServerStatusHandler 获取媒体服务器状态
|
||||
// GET /api/media/status
|
||||
func GetMediaServerStatusHandler(c *gin.Context) {
|
||||
ms := mediaserver.GetServer()
|
||||
config := ms.GetConfig()
|
||||
|
||||
if config == nil {
|
||||
utils.Error(c, 503, "媒体服务未配置")
|
||||
return
|
||||
}
|
||||
|
||||
response := MediaServerStatusResponse{
|
||||
Enabled: config.Enabled,
|
||||
RoomCount: ms.GetRoomCount(),
|
||||
Config: map[string]interface{}{
|
||||
"public_ip": config.PublicIP,
|
||||
"webrtc_port": config.WebRTCPort,
|
||||
"rtmp_port": config.RTMPPort,
|
||||
"http_flv_port": config.HTTPFLVPort,
|
||||
"max_room_size": config.MaxRoomSize,
|
||||
},
|
||||
}
|
||||
|
||||
// 获取 SFU 轨道数
|
||||
if sfu := ms.GetSFU(); sfu != nil {
|
||||
response.SFUTrackCount = sfu.GetTrackCount()
|
||||
}
|
||||
|
||||
// 获取 RTMP 流数
|
||||
if rtmp := ms.GetRTMP(); rtmp != nil {
|
||||
response.RTMPStreamCount = rtmp.GetStreamCount()
|
||||
}
|
||||
|
||||
utils.SuccessWithData(c, response, "获取成功")
|
||||
}
|
||||
|
||||
// GetAllRoomsHandler 获取所有房间
|
||||
// GET /api/media/rooms
|
||||
func GetAllRoomsHandler(c *gin.Context) {
|
||||
ms := mediaserver.GetServer()
|
||||
rooms := ms.GetAllRooms()
|
||||
|
||||
utils.SuccessWithData(c, gin.H{
|
||||
"count": len(rooms),
|
||||
"rooms": rooms,
|
||||
}, "获取成功")
|
||||
}
|
||||
|
||||
// GetRoomDetailHandler 获取房间详情
|
||||
// GET /api/media/rooms/:room_id
|
||||
func GetRoomDetailHandler(c *gin.Context) {
|
||||
roomID := c.Param("room_id")
|
||||
if roomID == "" {
|
||||
utils.BadRequest(c, "房间ID不能为空")
|
||||
return
|
||||
}
|
||||
|
||||
ms := mediaserver.GetServer()
|
||||
room := ms.GetRoom(roomID)
|
||||
if room == nil {
|
||||
utils.NotFound(c, "房间不存在")
|
||||
return
|
||||
}
|
||||
|
||||
info := room.GetInfo()
|
||||
|
||||
// 获取桥接信息
|
||||
bm := mediaserver.GetBridgeManager()
|
||||
bridge := bm.GetBridge(roomID)
|
||||
var bridgeInfo map[string]interface{}
|
||||
if bridge != nil && bridge.IsRunning() {
|
||||
w2r, r2w := bridge.GetAllConverters()
|
||||
w2rCount, r2wCount := bridge.GetConverterCount()
|
||||
bridgeInfo = map[string]interface{}{
|
||||
"running": true,
|
||||
"webrtc_to_rtmp": w2rCount,
|
||||
"rtmp_to_webrtc": r2wCount,
|
||||
"w2r_converters": w2r,
|
||||
"r2w_converters": r2w,
|
||||
}
|
||||
}
|
||||
|
||||
utils.SuccessWithData(c, gin.H{
|
||||
"room": info,
|
||||
"bridge": bridgeInfo,
|
||||
}, "获取成功")
|
||||
}
|
||||
|
||||
// GetRTMPStreamsHandler 获取 RTMP 流列表
|
||||
// GET /api/media/rtmp/streams
|
||||
func GetRTMPStreamsHandler(c *gin.Context) {
|
||||
ms := mediaserver.GetServer()
|
||||
rtmp := ms.GetRTMP()
|
||||
|
||||
if rtmp == nil {
|
||||
utils.Error(c, 503, "RTMP服务未启用")
|
||||
return
|
||||
}
|
||||
|
||||
roomID := c.Query("room_id")
|
||||
|
||||
var streams []*mediaserver.RTMPStream
|
||||
if roomID != "" {
|
||||
streams = rtmp.GetStreamsByRoom(roomID)
|
||||
} else {
|
||||
// 获取所有流（简化实现：只返回数量）
|
||||
utils.SuccessWithData(c, gin.H{
|
||||
"count": rtmp.GetStreamCount(),
|
||||
}, "获取成功")
|
||||
return
|
||||
}
|
||||
|
||||
utils.SuccessWithData(c, gin.H{
|
||||
"count": len(streams),
|
||||
"streams": streams,
|
||||
}, "获取成功")
|
||||
}
|
||||
|
||||
// CloseRoomHandler 关闭房间
|
||||
// POST /api/media/rooms/:room_id/close
|
||||
func CloseRoomHandler(c *gin.Context) {
|
||||
roomID := c.Param("room_id")
|
||||
if roomID == "" {
|
||||
utils.BadRequest(c, "房间ID不能为空")
|
||||
return
|
||||
}
|
||||
|
||||
ms := mediaserver.GetServer()
|
||||
room := ms.GetRoom(roomID)
|
||||
if room == nil {
|
||||
utils.NotFound(c, "房间不存在")
|
||||
return
|
||||
}
|
||||
|
||||
// 移除桥接
|
||||
bm := mediaserver.GetBridgeManager()
|
||||
bm.RemoveBridge(roomID)
|
||||
|
||||
// 移除房间
|
||||
ms.RemoveRoom(roomID)
|
||||
|
||||
utils.Success(c, "房间已关闭")
|
||||
}
|
||||
|
||||
// RegisterMediaRoutes 注册媒体相关路由
|
||||
func RegisterMediaRoutes(router *gin.RouterGroup) {
|
||||
mediaGroup := router.Group("/media")
|
||||
{
|
||||
mediaGroup.GET("/status", GetMediaServerStatusHandler)
|
||||
mediaGroup.GET("/rooms", GetAllRoomsHandler)
|
||||
mediaGroup.GET("/rooms/:room_id", GetRoomDetailHandler)
|
||||
mediaGroup.POST("/rooms/:room_id/close", CloseRoomHandler)
|
||||
mediaGroup.GET("/rtmp/streams", GetRTMPStreamsHandler)
|
||||
}
|
||||
}
|
||||
|
||||
482
internal/mediaserver/bridge.go
Normal file
482
internal/mediaserver/bridge.go
Normal file
@@ -0,0 +1,482 @@
|
||||
/**
|
||||
* package mediaserver
|
||||
*
|
||||
* WebRTC ↔ RTMP 媒体桥接
|
||||
* 功能：
|
||||
* 1. WebRTC 流转 RTMP（H5用户 -> 小程序用户）
|
||||
* 2. RTMP 流转 WebRTC（小程序用户 -> H5用户）
|
||||
* 3. 音频转码：Opus ↔ AAC
|
||||
* 4. 视频透传：H.264
|
||||
*/
|
||||
package mediaserver
|
||||
|
||||
import (
|
||||
"fmt"
|
||||
"log"
|
||||
"sync"
|
||||
"time"
|
||||
|
||||
"github.com/pion/rtp"
|
||||
"github.com/pion/webrtc/v3"
|
||||
)
|
||||
|
||||
// Bridge WebRTC-RTMP 媒体桥
|
||||
type Bridge struct {
|
||||
roomID string
|
||||
mu sync.RWMutex
|
||||
running bool
|
||||
stopChan chan struct{}
|
||||
|
||||
// WebRTC -> RTMP 转换器
|
||||
webrtcToRTMP map[string]*WebRTCToRTMPConverter
|
||||
|
||||
// RTMP -> WebRTC 转换器
|
||||
rtmpToWebRTC map[string]*RTMPToWebRTCConverter
|
||||
}
|
||||
|
||||
// WebRTCToRTMPConverter WebRTC 转 RTMP 转换器
|
||||
type WebRTCToRTMPConverter struct {
|
||||
UserID string
|
||||
StreamID string
|
||||
RoomID string
|
||||
Running bool
|
||||
stopChan chan struct{}
|
||||
|
||||
// 输入: WebRTC Track
|
||||
VideoTrack *webrtc.TrackRemote
|
||||
AudioTrack *webrtc.TrackRemote
|
||||
|
||||
// 输出: RTMP 流信息
|
||||
OutputURL string // RTMP 推流地址
|
||||
|
||||
// 编解码器
|
||||
videoCodec string // h264
|
||||
audioCodec string // opus -> aac
|
||||
|
||||
// 缓冲区
|
||||
videoBuffer chan *rtp.Packet
|
||||
audioBuffer chan *rtp.Packet
|
||||
}
|
||||
|
||||
// RTMPToWebRTCConverter RTMP 转 WebRTC 转换器
|
||||
type RTMPToWebRTCConverter struct {
|
||||
UserID string
|
||||
StreamID string
|
||||
RoomID string
|
||||
Running bool
|
||||
stopChan chan struct{}
|
||||
|
||||
// 输入: RTMP 拉流地址
|
||||
InputURL string
|
||||
|
||||
// 输出: WebRTC Track
|
||||
VideoTrack *webrtc.TrackLocalStaticRTP
|
||||
AudioTrack *webrtc.TrackLocalStaticRTP
|
||||
|
||||
// 编解码器
|
||||
videoCodec string // h264
|
||||
audioCodec string // aac -> opus
|
||||
}
|
||||
|
||||
// BridgeManager 桥接管理器
|
||||
type BridgeManager struct {
|
||||
bridges map[string]*Bridge // roomID -> bridge
|
||||
mu sync.RWMutex
|
||||
}
|
||||
|
||||
// 全局桥接管理器
|
||||
var bridgeManager *BridgeManager
|
||||
var bridgeOnce sync.Once
|
||||
|
||||
// GetBridgeManager 获取桥接管理器单例
|
||||
func GetBridgeManager() *BridgeManager {
|
||||
bridgeOnce.Do(func() {
|
||||
bridgeManager = &BridgeManager{
|
||||
bridges: make(map[string]*Bridge),
|
||||
}
|
||||
})
|
||||
return bridgeManager
|
||||
}
|
||||
|
||||
// GetOrCreateBridge 获取或创建房间的桥接
|
||||
func (bm *BridgeManager) GetOrCreateBridge(roomID string) *Bridge {
|
||||
bm.mu.Lock()
|
||||
defer bm.mu.Unlock()
|
||||
|
||||
if bridge, exists := bm.bridges[roomID]; exists {
|
||||
return bridge
|
||||
}
|
||||
|
||||
bridge := NewBridge(roomID)
|
||||
bm.bridges[roomID] = bridge
|
||||
return bridge
|
||||
}
|
||||
|
||||
// RemoveBridge 移除房间的桥接
|
||||
func (bm *BridgeManager) RemoveBridge(roomID string) {
|
||||
bm.mu.Lock()
|
||||
defer bm.mu.Unlock()
|
||||
|
||||
if bridge, exists := bm.bridges[roomID]; exists {
|
||||
bridge.Stop()
|
||||
delete(bm.bridges, roomID)
|
||||
}
|
||||
}
|
||||
|
||||
// GetBridge 获取桥接
|
||||
func (bm *BridgeManager) GetBridge(roomID string) *Bridge {
|
||||
bm.mu.RLock()
|
||||
defer bm.mu.RUnlock()
|
||||
return bm.bridges[roomID]
|
||||
}
|
||||
|
||||
// NewBridge 创建新桥接
|
||||
func NewBridge(roomID string) *Bridge {
|
||||
return &Bridge{
|
||||
roomID: roomID,
|
||||
stopChan: make(chan struct{}),
|
||||
webrtcToRTMP: make(map[string]*WebRTCToRTMPConverter),
|
||||
rtmpToWebRTC: make(map[string]*RTMPToWebRTCConverter),
|
||||
}
|
||||
}
|
||||
|
||||
// Start 启动桥接
|
||||
func (b *Bridge) Start() {
|
||||
b.mu.Lock()
|
||||
defer b.mu.Unlock()
|
||||
|
||||
if b.running {
|
||||
return
|
||||
}
|
||||
b.running = true
|
||||
log.Printf("🌉 [Bridge] 启动房间桥接 | Room:%s", b.roomID)
|
||||
}
|
||||
|
||||
// Stop 停止桥接
|
||||
func (b *Bridge) Stop() {
|
||||
b.mu.Lock()
|
||||
defer b.mu.Unlock()
|
||||
|
||||
if !b.running {
|
||||
return
|
||||
}
|
||||
|
||||
// 关闭停止信号
|
||||
close(b.stopChan)
|
||||
|
||||
// 停止所有 WebRTC -> RTMP 转换器
|
||||
for _, converter := range b.webrtcToRTMP {
|
||||
converter.Stop()
|
||||
}
|
||||
b.webrtcToRTMP = make(map[string]*WebRTCToRTMPConverter)
|
||||
|
||||
// 停止所有 RTMP -> WebRTC 转换器
|
||||
for _, converter := range b.rtmpToWebRTC {
|
||||
converter.Stop()
|
||||
}
|
||||
b.rtmpToWebRTC = make(map[string]*RTMPToWebRTCConverter)
|
||||
|
||||
b.running = false
|
||||
log.Printf("🌉 [Bridge] 停止房间桥接 | Room:%s", b.roomID)
|
||||
}
|
||||
|
||||
// AddWebRTCToRTMP 添加 WebRTC 到 RTMP 的转换
|
||||
func (b *Bridge) AddWebRTCToRTMP(userID string, videoTrack, audioTrack *webrtc.TrackRemote, outputURL string) (*WebRTCToRTMPConverter, error) {
|
||||
b.mu.Lock()
|
||||
defer b.mu.Unlock()
|
||||
|
||||
if !b.running {
|
||||
b.running = true
|
||||
}
|
||||
|
||||
streamID := fmt.Sprintf("w2r_%s_%s_%d", b.roomID, userID, time.Now().UnixNano())
|
||||
|
||||
converter := &WebRTCToRTMPConverter{
|
||||
UserID: userID,
|
||||
StreamID: streamID,
|
||||
RoomID: b.roomID,
|
||||
Running: true,
|
||||
stopChan: make(chan struct{}),
|
||||
VideoTrack: videoTrack,
|
||||
AudioTrack: audioTrack,
|
||||
OutputURL: outputURL,
|
||||
videoCodec: "h264",
|
||||
audioCodec: "opus",
|
||||
videoBuffer: make(chan *rtp.Packet, 100),
|
||||
audioBuffer: make(chan *rtp.Packet, 100),
|
||||
}
|
||||
|
||||
b.webrtcToRTMP[userID] = converter
|
||||
|
||||
// 启动转换协程
|
||||
go converter.Run()
|
||||
|
||||
log.Printf("🌉 [Bridge] 添加 WebRTC->RTMP | Room:%s User:%s Stream:%s", b.roomID, userID, streamID)
|
||||
return converter, nil
|
||||
}
|
||||
|
||||
// AddRTMPToWebRTC 添加 RTMP 到 WebRTC 的转换
|
||||
func (b *Bridge) AddRTMPToWebRTC(userID string, inputURL string) (*RTMPToWebRTCConverter, error) {
|
||||
b.mu.Lock()
|
||||
defer b.mu.Unlock()
|
||||
|
||||
if !b.running {
|
||||
b.running = true
|
||||
}
|
||||
|
||||
streamID := fmt.Sprintf("r2w_%s_%s_%d", b.roomID, userID, time.Now().UnixNano())
|
||||
|
||||
// 创建本地轨道
|
||||
videoTrack, err := webrtc.NewTrackLocalStaticRTP(
|
||||
webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH264},
|
||||
fmt.Sprintf("video_%s", streamID),
|
||||
fmt.Sprintf("stream_%s", userID),
|
||||
)
|
||||
if err != nil {
|
||||
return nil, fmt.Errorf("create video track failed: %w", err)
|
||||
}
|
||||
|
||||
audioTrack, err := webrtc.NewTrackLocalStaticRTP(
|
||||
webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus},
|
||||
fmt.Sprintf("audio_%s", streamID),
|
||||
fmt.Sprintf("stream_%s", userID),
|
||||
)
|
||||
if err != nil {
|
||||
return nil, fmt.Errorf("create audio track failed: %w", err)
|
||||
}
|
||||
|
||||
converter := &RTMPToWebRTCConverter{
|
||||
UserID: userID,
|
||||
StreamID: streamID,
|
||||
RoomID: b.roomID,
|
||||
Running: true,
|
||||
stopChan: make(chan struct{}),
|
||||
InputURL: inputURL,
|
||||
VideoTrack: videoTrack,
|
||||
AudioTrack: audioTrack,
|
||||
videoCodec: "h264",
|
||||
audioCodec: "aac",
|
||||
}
|
||||
|
||||
b.rtmpToWebRTC[userID] = converter
|
||||
|
||||
// 启动转换协程
|
||||
go converter.Run()
|
||||
|
||||
log.Printf("🌉 [Bridge] 添加 RTMP->WebRTC | Room:%s User:%s Stream:%s", b.roomID, userID, streamID)
|
||||
return converter, nil
|
||||
}
|
||||
|
||||
// RemoveWebRTCToRTMP 移除 WebRTC 到 RTMP 的转换
|
||||
func (b *Bridge) RemoveWebRTCToRTMP(userID string) {
|
||||
b.mu.Lock()
|
||||
defer b.mu.Unlock()
|
||||
|
||||
if converter, exists := b.webrtcToRTMP[userID]; exists {
|
||||
converter.Stop()
|
||||
delete(b.webrtcToRTMP, userID)
|
||||
log.Printf("🌉 [Bridge] 移除 WebRTC->RTMP | Room:%s User:%s", b.roomID, userID)
|
||||
}
|
||||
}
|
||||
|
||||
// RemoveRTMPToWebRTC 移除 RTMP 到 WebRTC 的转换
|
||||
func (b *Bridge) RemoveRTMPToWebRTC(userID string) {
|
||||
b.mu.Lock()
|
||||
defer b.mu.Unlock()
|
||||
|
||||
if converter, exists := b.rtmpToWebRTC[userID]; exists {
|
||||
converter.Stop()
|
||||
delete(b.rtmpToWebRTC, userID)
|
||||
log.Printf("🌉 [Bridge] 移除 RTMP->WebRTC | Room:%s User:%s", b.roomID, userID)
|
||||
}
|
||||
}
|
||||
|
||||
// GetRTMPToWebRTCTracks 获取 RTMP->WebRTC 转换器的输出轨道
|
||||
func (b *Bridge) GetRTMPToWebRTCTracks(userID string) (*webrtc.TrackLocalStaticRTP, *webrtc.TrackLocalStaticRTP) {
|
||||
b.mu.RLock()
|
||||
defer b.mu.RUnlock()
|
||||
|
||||
if converter, exists := b.rtmpToWebRTC[userID]; exists {
|
||||
return converter.VideoTrack, converter.AudioTrack
|
||||
}
|
||||
return nil, nil
|
||||
}
|
||||
|
||||
// --- WebRTCToRTMPConverter 方法 ---
|
||||
|
||||
// Run 执行 WebRTC 到 RTMP 的转换
|
||||
func (c *WebRTCToRTMPConverter) Run() {
|
||||
log.Printf("🌉 [W2R] 转换开始 | Stream:%s", c.StreamID)
|
||||
defer log.Printf("🌉 [W2R] 转换结束 | Stream:%s", c.StreamID)
|
||||
|
||||
// 启动视频读取
|
||||
if c.VideoTrack != nil {
|
||||
go c.readVideoTrack()
|
||||
}
|
||||
|
||||
// 启动音频读取
|
||||
if c.AudioTrack != nil {
|
||||
go c.readAudioTrack()
|
||||
}
|
||||
|
||||
// 启动 RTMP 推流
|
||||
go c.pushToRTMP()
|
||||
|
||||
// 等待停止信号
|
||||
<-c.stopChan
|
||||
}
|
||||
|
||||
// readVideoTrack 读取视频轨道
|
||||
func (c *WebRTCToRTMPConverter) readVideoTrack() {
|
||||
buf := make([]byte, 1500)
|
||||
for c.Running {
|
||||
n, _, err := c.VideoTrack.Read(buf)
|
||||
if err != nil {
|
||||
if c.Running {
|
||||
log.Printf("⚠️ [W2R] 读取视频失败: %v", err)
|
||||
}
|
||||
return
|
||||
}
|
||||
|
||||
// 解析 RTP 包
|
||||
packet := &rtp.Packet{}
|
||||
if err := packet.Unmarshal(buf[:n]); err != nil {
|
||||
continue
|
||||
}
|
||||
|
||||
select {
|
||||
case c.videoBuffer <- packet:
|
||||
default:
|
||||
// 缓冲区满，丢弃
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// readAudioTrack 读取音频轨道
|
||||
func (c *WebRTCToRTMPConverter) readAudioTrack() {
|
||||
buf := make([]byte, 1500)
|
||||
for c.Running {
|
||||
n, _, err := c.AudioTrack.Read(buf)
|
||||
if err != nil {
|
||||
if c.Running {
|
||||
log.Printf("⚠️ [W2R] 读取音频失败: %v", err)
|
||||
}
|
||||
return
|
||||
}
|
||||
|
||||
// 解析 RTP 包
|
||||
packet := &rtp.Packet{}
|
||||
if err := packet.Unmarshal(buf[:n]); err != nil {
|
||||
continue
|
||||
}
|
||||
|
||||
select {
|
||||
case c.audioBuffer <- packet:
|
||||
default:
|
||||
// 缓冲区满，丢弃
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// pushToRTMP 推送到 RTMP
|
||||
func (c *WebRTCToRTMPConverter) pushToRTMP() {
|
||||
// TODO: 实际的 RTMP 推流实现
|
||||
// 1. 连接到 RTMP 服务器
|
||||
// 2. 从 videoBuffer/audioBuffer 读取 RTP 包
|
||||
// 3. 解封装 RTP，获取裸数据
|
||||
// 4. 视频: H.264 NAL 单元 -> FLV Tag
|
||||
// 5. 音频: Opus -> AAC (需要转码)
|
||||
// 6. 发送 FLV Tag 到 RTMP 服务器
|
||||
|
||||
for c.Running {
|
||||
select {
|
||||
case <-c.stopChan:
|
||||
return
|
||||
case videoPacket := <-c.videoBuffer:
|
||||
// 处理视频包
|
||||
_ = videoPacket
|
||||
case audioPacket := <-c.audioBuffer:
|
||||
// 处理音频包
|
||||
_ = audioPacket
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// Stop 停止转换
|
||||
func (c *WebRTCToRTMPConverter) Stop() {
|
||||
if !c.Running {
|
||||
return
|
||||
}
|
||||
c.Running = false
|
||||
close(c.stopChan)
|
||||
}
|
||||
|
||||
// --- RTMPToWebRTCConverter 方法 ---
|
||||
|
||||
// Run 执行 RTMP 到 WebRTC 的转换
|
||||
func (c *RTMPToWebRTCConverter) Run() {
|
||||
log.Printf("🌉 [R2W] 转换开始 | Stream:%s URL:%s", c.StreamID, c.InputURL)
|
||||
defer log.Printf("🌉 [R2W] 转换结束 | Stream:%s", c.StreamID)
|
||||
|
||||
// TODO: 实际的 RTMP 拉流实现
|
||||
// 1. 连接到 RTMP 服务器
|
||||
// 2. 拉取 FLV 流
|
||||
// 3. 解析 FLV Tag
|
||||
// 4. 视频: H.264 NAL 单元 -> RTP 包
|
||||
// 5. 音频: AAC -> Opus (需要转码)
|
||||
// 6. 写入 WebRTC Track
|
||||
|
||||
// 等待停止信号
|
||||
<-c.stopChan
|
||||
}
|
||||
|
||||
// Stop 停止转换
|
||||
func (c *RTMPToWebRTCConverter) Stop() {
|
||||
if !c.Running {
|
||||
return
|
||||
}
|
||||
c.Running = false
|
||||
close(c.stopChan)
|
||||
}
|
||||
|
||||
// GetConverterCount 获取转换器数量
|
||||
func (b *Bridge) GetConverterCount() (int, int) {
|
||||
b.mu.RLock()
|
||||
defer b.mu.RUnlock()
|
||||
return len(b.webrtcToRTMP), len(b.rtmpToWebRTC)
|
||||
}
|
||||
|
||||
// IsRunning 是否运行中
|
||||
func (b *Bridge) IsRunning() bool {
|
||||
b.mu.RLock()
|
||||
defer b.mu.RUnlock()
|
||||
return b.running
|
||||
}
|
||||
|
||||
// GetAllConverters 获取所有转换器信息
|
||||
func (b *Bridge) GetAllConverters() ([]map[string]interface{}, []map[string]interface{}) {
|
||||
b.mu.RLock()
|
||||
defer b.mu.RUnlock()
|
||||
|
||||
w2r := make([]map[string]interface{}, 0, len(b.webrtcToRTMP))
|
||||
for _, c := range b.webrtcToRTMP {
|
||||
w2r = append(w2r, map[string]interface{}{
|
||||
"user_id": c.UserID,
|
||||
"stream_id": c.StreamID,
|
||||
"output_url": c.OutputURL,
|
||||
"running": c.Running,
|
||||
})
|
||||
}
|
||||
|
||||
r2w := make([]map[string]interface{}, 0, len(b.rtmpToWebRTC))
|
||||
for _, c := range b.rtmpToWebRTC {
|
||||
r2w = append(r2w, map[string]interface{}{
|
||||
"user_id": c.UserID,
|
||||
"stream_id": c.StreamID,
|
||||
"input_url": c.InputURL,
|
||||
"running": c.Running,
|
||||
})
|
||||
}
|
||||
|
||||
return w2r, r2w
|
||||
}
|
||||
35
internal/mediaserver/errors.go
Normal file
35
internal/mediaserver/errors.go
Normal file
@@ -0,0 +1,35 @@
|
||||
/**
|
||||
* package mediaserver
|
||||
*
|
||||
* 错误定义
|
||||
*/
|
||||
package mediaserver
|
||||
|
||||
import "errors"
|
||||
|
||||
var (
|
||||
// 房间相关错误
|
||||
ErrRoomNotFound = errors.New("room not found")
|
||||
ErrRoomFull = errors.New("room is full")
|
||||
ErrRoomClosed = errors.New("room is closed")
|
||||
ErrAlreadyInRoom = errors.New("user already in room")
|
||||
ErrNotInRoom = errors.New("user not in room")
|
||||
|
||||
// 参与者相关错误
|
||||
ErrParticipantNotFound = errors.New("participant not found")
|
||||
ErrInvalidParticipantType = errors.New("invalid participant type")
|
||||
|
||||
// 媒体相关错误
|
||||
ErrMediaServerDisabled = errors.New("media server is disabled")
|
||||
ErrSFUNotReady = errors.New("SFU server is not ready")
|
||||
ErrRTMPNotReady = errors.New("RTMP server is not ready")
|
||||
ErrStreamNotFound = errors.New("stream not found")
|
||||
ErrInvalidOffer = errors.New("invalid SDP offer")
|
||||
ErrInvalidAnswer = errors.New("invalid SDP answer")
|
||||
ErrICEConnectionFailed = errors.New("ICE connection failed")
|
||||
|
||||
// 编解码相关错误
|
||||
ErrUnsupportedCodec = errors.New("unsupported codec")
|
||||
ErrTranscodeFailed = errors.New("transcode failed")
|
||||
)
|
||||
|
||||
299
internal/mediaserver/room.go
Normal file
299
internal/mediaserver/room.go
Normal file
@@ -0,0 +1,299 @@
|
||||
/**
|
||||
* package mediaserver
|
||||
*
|
||||
* 通话房间管理
|
||||
* 功能：
|
||||
* 1. 管理房间内的参与者
|
||||
* 2. 管理 WebRTC 和 RTMP 流
|
||||
* 3. 处理用户加入/离开
|
||||
*/
|
||||
package mediaserver
|
||||
|
||||
import (
|
||||
"log"
|
||||
"sync"
|
||||
"time"
|
||||
)
|
||||
|
||||
// ParticipantType 参与者类型
|
||||
type ParticipantType int
|
||||
|
||||
const (
|
||||
ParticipantTypeWebRTC ParticipantType = iota // H5/App WebRTC 用户
|
||||
ParticipantTypeRTMP // 小程序 RTMP 用户
|
||||
)
|
||||
|
||||
// Participant 房间参与者
|
||||
type Participant struct {
|
||||
UserID string `json:"user_id"`
|
||||
Type ParticipantType `json:"type"`
|
||||
JoinedAt time.Time `json:"joined_at"`
|
||||
HasAudio bool `json:"has_audio"`
|
||||
HasVideo bool `json:"has_video"`
|
||||
|
||||
// WebRTC 相关
|
||||
PeerConnection interface{} `json:"-"` // *webrtc.PeerConnection
|
||||
|
||||
// RTMP 相关
|
||||
PushURL string `json:"push_url,omitempty"` // 推流地址
|
||||
PullURL string `json:"pull_url,omitempty"` // 拉流地址
|
||||
StreamID string `json:"stream_id,omitempty"` // 流ID
|
||||
}
|
||||
|
||||
// Room 通话房间
|
||||
type Room struct {
|
||||
ID string `json:"id"`
|
||||
CreatedAt time.Time `json:"created_at"`
|
||||
MaxSize int `json:"max_size"`
|
||||
Participants map[string]*Participant `json:"participants"`
|
||||
|
||||
// 房间状态
|
||||
CallType string `json:"call_type"` // audio/video
|
||||
IsGroupCall bool `json:"is_group_call"`
|
||||
|
||||
mu sync.RWMutex
|
||||
closeChan chan struct{}
|
||||
closed bool
|
||||
}
|
||||
|
||||
// RoomInfo 房间信息（用于API返回）
|
||||
type RoomInfo struct {
|
||||
ID string `json:"id"`
|
||||
CreatedAt time.Time `json:"created_at"`
|
||||
ParticipantCount int `json:"participant_count"`
|
||||
CallType string `json:"call_type"`
|
||||
IsGroupCall bool `json:"is_group_call"`
|
||||
Participants []ParticipantInfo `json:"participants"`
|
||||
}
|
||||
|
||||
// ParticipantInfo 参与者信息（用于API返回）
|
||||
type ParticipantInfo struct {
|
||||
UserID string `json:"user_id"`
|
||||
Type ParticipantType `json:"type"`
|
||||
TypeName string `json:"type_name"`
|
||||
HasAudio bool `json:"has_audio"`
|
||||
HasVideo bool `json:"has_video"`
|
||||
JoinedAt time.Time `json:"joined_at"`
|
||||
}
|
||||
|
||||
// NewRoom 创建新房间
|
||||
func NewRoom(id string, maxSize int) *Room {
|
||||
return &Room{
|
||||
ID: id,
|
||||
CreatedAt: time.Now(),
|
||||
MaxSize: maxSize,
|
||||
Participants: make(map[string]*Participant),
|
||||
closeChan: make(chan struct{}),
|
||||
}
|
||||
}
|
||||
|
||||
// SetCallType 设置通话类型
|
||||
func (r *Room) SetCallType(callType string, isGroup bool) {
|
||||
r.mu.Lock()
|
||||
defer r.mu.Unlock()
|
||||
r.CallType = callType
|
||||
r.IsGroupCall = isGroup
|
||||
}
|
||||
|
||||
// AddParticipant 添加参与者
|
||||
func (r *Room) AddParticipant(userID string, pType ParticipantType) (*Participant, error) {
|
||||
r.mu.Lock()
|
||||
defer r.mu.Unlock()
|
||||
|
||||
if r.closed {
|
||||
return nil, ErrRoomClosed
|
||||
}
|
||||
|
||||
// 检查是否已存在
|
||||
if p, exists := r.Participants[userID]; exists {
|
||||
return p, nil
|
||||
}
|
||||
|
||||
// 检查房间容量
|
||||
if len(r.Participants) >= r.MaxSize {
|
||||
return nil, ErrRoomFull
|
||||
}
|
||||
|
||||
participant := &Participant{
|
||||
UserID: userID,
|
||||
Type: pType,
|
||||
JoinedAt: time.Now(),
|
||||
}
|
||||
|
||||
r.Participants[userID] = participant
|
||||
log.Printf("🎥 [Room:%s] 用户 %s 加入 (类型: %d)", r.ID, userID, pType)
|
||||
|
||||
return participant, nil
|
||||
}
|
||||
|
||||
// RemoveParticipant 移除参与者
|
||||
func (r *Room) RemoveParticipant(userID string) {
|
||||
r.mu.Lock()
|
||||
defer r.mu.Unlock()
|
||||
|
||||
if p, exists := r.Participants[userID]; exists {
|
||||
// 清理 PeerConnection
|
||||
if p.PeerConnection != nil {
|
||||
// TODO: 关闭 PeerConnection
|
||||
}
|
||||
delete(r.Participants, userID)
|
||||
log.Printf("🎥 [Room:%s] 用户 %s 离开", r.ID, userID)
|
||||
}
|
||||
|
||||
// 如果房间空了，标记为可清理
|
||||
if len(r.Participants) == 0 {
|
||||
log.Printf("🎥 [Room:%s] 房间已空，等待清理", r.ID)
|
||||
}
|
||||
}
|
||||
|
||||
// GetParticipant 获取参与者
|
||||
func (r *Room) GetParticipant(userID string) *Participant {
|
||||
r.mu.RLock()
|
||||
defer r.mu.RUnlock()
|
||||
return r.Participants[userID]
|
||||
}
|
||||
|
||||
// GetAllParticipants 获取所有参与者
|
||||
func (r *Room) GetAllParticipants() []*Participant {
|
||||
r.mu.RLock()
|
||||
defer r.mu.RUnlock()
|
||||
|
||||
participants := make([]*Participant, 0, len(r.Participants))
|
||||
for _, p := range r.Participants {
|
||||
participants = append(participants, p)
|
||||
}
|
||||
return participants
|
||||
}
|
||||
|
||||
// GetOtherParticipants 获取除指定用户外的其他参与者
|
||||
func (r *Room) GetOtherParticipants(excludeUserID string) []*Participant {
|
||||
r.mu.RLock()
|
||||
defer r.mu.RUnlock()
|
||||
|
||||
participants := make([]*Participant, 0, len(r.Participants)-1)
|
||||
for _, p := range r.Participants {
|
||||
if p.UserID != excludeUserID {
|
||||
participants = append(participants, p)
|
||||
}
|
||||
}
|
||||
return participants
|
||||
}
|
||||
|
||||
// GetParticipantCount 获取参与者数量
|
||||
func (r *Room) GetParticipantCount() int {
|
||||
r.mu.RLock()
|
||||
defer r.mu.RUnlock()
|
||||
return len(r.Participants)
|
||||
}
|
||||
|
||||
// IsEmpty 房间是否为空
|
||||
func (r *Room) IsEmpty() bool {
|
||||
r.mu.RLock()
|
||||
defer r.mu.RUnlock()
|
||||
return len(r.Participants) == 0
|
||||
}
|
||||
|
||||
// IsFull 房间是否已满
|
||||
func (r *Room) IsFull() bool {
|
||||
r.mu.RLock()
|
||||
defer r.mu.RUnlock()
|
||||
return len(r.Participants) >= r.MaxSize
|
||||
}
|
||||
|
||||
// GetInfo 获取房间信息
|
||||
func (r *Room) GetInfo() *RoomInfo {
|
||||
r.mu.RLock()
|
||||
defer r.mu.RUnlock()
|
||||
|
||||
participants := make([]ParticipantInfo, 0, len(r.Participants))
|
||||
for _, p := range r.Participants {
|
||||
typeName := "WebRTC"
|
||||
if p.Type == ParticipantTypeRTMP {
|
||||
typeName = "RTMP"
|
||||
}
|
||||
participants = append(participants, ParticipantInfo{
|
||||
UserID: p.UserID,
|
||||
Type: p.Type,
|
||||
TypeName: typeName,
|
||||
HasAudio: p.HasAudio,
|
||||
HasVideo: p.HasVideo,
|
||||
JoinedAt: p.JoinedAt,
|
||||
})
|
||||
}
|
||||
|
||||
return &RoomInfo{
|
||||
ID: r.ID,
|
||||
CreatedAt: r.CreatedAt,
|
||||
ParticipantCount: len(r.Participants),
|
||||
CallType: r.CallType,
|
||||
IsGroupCall: r.IsGroupCall,
|
||||
Participants: participants,
|
||||
}
|
||||
}
|
||||
|
||||
// UpdateParticipantMedia 更新参与者媒体状态
|
||||
func (r *Room) UpdateParticipantMedia(userID string, hasAudio, hasVideo bool) {
|
||||
r.mu.Lock()
|
||||
defer r.mu.Unlock()
|
||||
|
||||
if p, exists := r.Participants[userID]; exists {
|
||||
p.HasAudio = hasAudio
|
||||
p.HasVideo = hasVideo
|
||||
}
|
||||
}
|
||||
|
||||
// SetParticipantRTMPURLs 设置参与者的RTMP推拉流地址
|
||||
func (r *Room) SetParticipantRTMPURLs(userID, pushURL, pullURL, streamID string) {
|
||||
r.mu.Lock()
|
||||
defer r.mu.Unlock()
|
||||
|
||||
if p, exists := r.Participants[userID]; exists {
|
||||
p.PushURL = pushURL
|
||||
p.PullURL = pullURL
|
||||
p.StreamID = streamID
|
||||
}
|
||||
}
|
||||
|
||||
// Close 关闭房间
|
||||
func (r *Room) Close() {
|
||||
r.mu.Lock()
|
||||
defer r.mu.Unlock()
|
||||
|
||||
if r.closed {
|
||||
return
|
||||
}
|
||||
|
||||
r.closed = true
|
||||
close(r.closeChan)
|
||||
|
||||
// 清理所有参与者
|
||||
for userID, p := range r.Participants {
|
||||
if p.PeerConnection != nil {
|
||||
// TODO: 关闭 PeerConnection
|
||||
}
|
||||
delete(r.Participants, userID)
|
||||
}
|
||||
|
||||
log.Printf("🎥 [Room:%s] 已关闭", r.ID)
|
||||
}
|
||||
|
||||
// IsClosed 房间是否已关闭
|
||||
func (r *Room) IsClosed() bool {
|
||||
r.mu.RLock()
|
||||
defer r.mu.RUnlock()
|
||||
return r.closed
|
||||
}
|
||||
|
||||
// Broadcast 向房间内所有参与者广播消息（用于信令）
|
||||
func (r *Room) Broadcast(excludeUserID string, message interface{}) {
|
||||
r.mu.RLock()
|
||||
defer r.mu.RUnlock()
|
||||
|
||||
for userID := range r.Participants {
|
||||
if userID != excludeUserID {
|
||||
// TODO: 通过 WebSocket 发送消息
|
||||
_ = message
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
501
internal/mediaserver/rtmp.go
Normal file
501
internal/mediaserver/rtmp.go
Normal file
@@ -0,0 +1,501 @@
|
||||
/**
|
||||
* package mediaserver
|
||||
*
|
||||
* RTMP 服务（用于微信小程序 live-pusher/live-player）
|
||||
* 功能：
|
||||
* 1. 接收小程序推流
|
||||
* 2. 提供拉流服务
|
||||
* 3. 支持 HTTP-FLV 协议
|
||||
*
|
||||
* 注意：当前版本使用简化实现，实际 RTMP 协议处理需要集成完整的 RTMP 服务器
|
||||
* 生产环境建议使用 nginx-rtmp 或独立部署 livego
|
||||
*/
|
||||
package mediaserver
|
||||
|
||||
import (
|
||||
"crypto/hmac"
|
||||
"crypto/sha1"
|
||||
"encoding/base64"
|
||||
"fmt"
|
||||
"io"
|
||||
"log"
|
||||
"net"
|
||||
"net/http"
|
||||
"sync"
|
||||
"time"
|
||||
|
||||
"github.com/spf13/viper"
|
||||
)
|
||||
|
||||
// RTMPServer RTMP 服务器
|
||||
type RTMPServer struct {
|
||||
config *MediaServerConfig
|
||||
mu sync.RWMutex
|
||||
running bool
|
||||
streams map[string]*RTMPStream // streamID -> stream
|
||||
|
||||
// 服务器监听器
|
||||
rtmpListener net.Listener
|
||||
httpServer *http.Server
|
||||
}
|
||||
|
||||
// RTMPStream RTMP 流信息
|
||||
type RTMPStream struct {
|
||||
ID string `json:"id"`
|
||||
RoomID string `json:"room_id"`
|
||||
UserID string `json:"user_id"`
|
||||
PushURL string `json:"push_url"`
|
||||
PullURL string `json:"pull_url"`
|
||||
FLVURL string `json:"flv_url"`
|
||||
CreatedAt time.Time `json:"created_at"`
|
||||
IsActive bool `json:"is_active"`
|
||||
|
||||
// 流数据缓存
|
||||
dataChan chan []byte `json:"-"`
|
||||
stopChan chan struct{} `json:"-"`
|
||||
}
|
||||
|
||||
// NewRTMPServer 创建 RTMP 服务器
|
||||
func NewRTMPServer(config *MediaServerConfig) *RTMPServer {
|
||||
return &RTMPServer{
|
||||
config: config,
|
||||
streams: make(map[string]*RTMPStream),
|
||||
}
|
||||
}
|
||||
|
||||
// Start 启动 RTMP 服务器
|
||||
func (r *RTMPServer) Start() {
|
||||
r.mu.Lock()
|
||||
if r.running {
|
||||
r.mu.Unlock()
|
||||
return
|
||||
}
|
||||
r.running = true
|
||||
r.mu.Unlock()
|
||||
|
||||
// 启动 RTMP 服务器（简化版）
|
||||
go r.startRTMPServer()
|
||||
|
||||
// 启动 HTTP-FLV 服务器
|
||||
go r.startHTTPFLVServer()
|
||||
|
||||
// 启动流清理任务
|
||||
go r.cleanupTask()
|
||||
|
||||
log.Printf("🚀 [RTMP] RTMP 服务已启动 | RTMP端口: %d | HTTP-FLV端口: %d",
|
||||
r.config.RTMPPort, r.config.HTTPFLVPort)
|
||||
}
|
||||
|
||||
// startRTMPServer 启动 RTMP 服务（简化版）
|
||||
func (r *RTMPServer) startRTMPServer() {
|
||||
addr := fmt.Sprintf(":%d", r.config.RTMPPort)
|
||||
|
||||
var err error
|
||||
r.rtmpListener, err = net.Listen("tcp", addr)
|
||||
if err != nil {
|
||||
log.Printf("❌ [RTMP] RTMP 监听失败: %v", err)
|
||||
return
|
||||
}
|
||||
|
||||
log.Printf("🎬 [RTMP] RTMP 服务监听: %s", addr)
|
||||
|
||||
for {
|
||||
r.mu.RLock()
|
||||
if !r.running {
|
||||
r.mu.RUnlock()
|
||||
return
|
||||
}
|
||||
r.mu.RUnlock()
|
||||
|
||||
conn, err := r.rtmpListener.Accept()
|
||||
if err != nil {
|
||||
if !r.running {
|
||||
return
|
||||
}
|
||||
log.Printf("⚠️ [RTMP] 接受连接失败: %v", err)
|
||||
continue
|
||||
}
|
||||
|
||||
// 处理 RTMP 连接
|
||||
go r.handleRTMPConnection(conn)
|
||||
}
|
||||
}
|
||||
|
||||
// handleRTMPConnection 处理 RTMP 连接
|
||||
// 简化版：基本的 RTMP 握手和流处理
|
||||
func (r *RTMPServer) handleRTMPConnection(conn net.Conn) {
|
||||
defer conn.Close()
|
||||
|
||||
// RTMP 协议处理
|
||||
// 步骤1: RTMP 握手
|
||||
if err := r.doHandshake(conn); err != nil {
|
||||
log.Printf("⚠️ [RTMP] 握手失败: %v", err)
|
||||
return
|
||||
}
|
||||
|
||||
log.Printf("🎬 [RTMP] 新连接: %s", conn.RemoteAddr().String())
|
||||
|
||||
// 步骤2: 读取 RTMP 消息
|
||||
// 简化实现：等待数据并转发
|
||||
buf := make([]byte, 4096)
|
||||
for {
|
||||
r.mu.RLock()
|
||||
if !r.running {
|
||||
r.mu.RUnlock()
|
||||
return
|
||||
}
|
||||
r.mu.RUnlock()
|
||||
|
||||
conn.SetReadDeadline(time.Now().Add(30 * time.Second))
|
||||
n, err := conn.Read(buf)
|
||||
if err != nil {
|
||||
if err != io.EOF {
|
||||
log.Printf("⚠️ [RTMP] 读取失败: %v", err)
|
||||
}
|
||||
return
|
||||
}
|
||||
|
||||
// TODO: 解析 RTMP 消息并路由到对应的流
|
||||
_ = n
|
||||
}
|
||||
}
|
||||
|
||||
// doHandshake 执行 RTMP 握手
|
||||
func (r *RTMPServer) doHandshake(conn net.Conn) error {
|
||||
// RTMP 握手协议
|
||||
// C0 + C1 -> S0 + S1 + S2 -> C2
|
||||
|
||||
c0c1 := make([]byte, 1537) // C0 (1) + C1 (1536)
|
||||
conn.SetReadDeadline(time.Now().Add(10 * time.Second))
|
||||
if _, err := io.ReadFull(conn, c0c1); err != nil {
|
||||
return fmt.Errorf("read C0C1 failed: %w", err)
|
||||
}
|
||||
|
||||
// 验证版本号 (C0 应该是 0x03)
|
||||
if c0c1[0] != 0x03 {
|
||||
return fmt.Errorf("invalid RTMP version: %d", c0c1[0])
|
||||
}
|
||||
|
||||
// 发送 S0 + S1 + S2
|
||||
s0s1s2 := make([]byte, 3073) // S0 (1) + S1 (1536) + S2 (1536)
|
||||
s0s1s2[0] = 0x03 // RTMP version 3
|
||||
|
||||
// S1: 时间戳(4) + 零(4) + 随机数据(1528)
|
||||
// S2: 客户端时间戳(4) + 服务器时间戳(4) + 随机数据(1528)
|
||||
copy(s0s1s2[1:1537], c0c1[1:]) // S1 = C1
|
||||
copy(s0s1s2[1537:], c0c1[1:]) // S2 = C1
|
||||
|
||||
conn.SetWriteDeadline(time.Now().Add(10 * time.Second))
|
||||
if _, err := conn.Write(s0s1s2); err != nil {
|
||||
return fmt.Errorf("write S0S1S2 failed: %w", err)
|
||||
}
|
||||
|
||||
// 读取 C2
|
||||
c2 := make([]byte, 1536)
|
||||
conn.SetReadDeadline(time.Now().Add(10 * time.Second))
|
||||
if _, err := io.ReadFull(conn, c2); err != nil {
|
||||
return fmt.Errorf("read C2 failed: %w", err)
|
||||
}
|
||||
|
||||
return nil
|
||||
}
|
||||
|
||||
// startHTTPFLVServer 启动 HTTP-FLV 服务
|
||||
func (r *RTMPServer) startHTTPFLVServer() {
|
||||
addr := fmt.Sprintf(":%d", r.config.HTTPFLVPort)
|
||||
|
||||
mux := http.NewServeMux()
|
||||
mux.HandleFunc("/live/", r.handleFLVRequest)
|
||||
mux.HandleFunc("/api/streams", r.handleStreamsAPI)
|
||||
|
||||
r.httpServer = &http.Server{
|
||||
Addr: addr,
|
||||
Handler: mux,
|
||||
}
|
||||
|
||||
log.Printf("🎬 [RTMP] HTTP-FLV 服务监听: %s", addr)
|
||||
|
||||
if err := r.httpServer.ListenAndServe(); err != nil && err != http.ErrServerClosed {
|
||||
log.Printf("❌ [RTMP] HTTP-FLV 启动失败: %v", err)
|
||||
}
|
||||
}
|
||||
|
||||
// handleFLVRequest 处理 FLV 请求
|
||||
func (r *RTMPServer) handleFLVRequest(w http.ResponseWriter, req *http.Request) {
|
||||
// 解析流ID: /live/{streamID}.flv
|
||||
path := req.URL.Path
|
||||
if len(path) < 7 { // "/live/" = 6, 至少还需要1个字符
|
||||
http.Error(w, "Invalid path", http.StatusBadRequest)
|
||||
return
|
||||
}
|
||||
|
||||
// 提取 streamID
|
||||
streamPath := path[6:] // 去掉 "/live/"
|
||||
if len(streamPath) > 4 && streamPath[len(streamPath)-4:] == ".flv" {
|
||||
streamPath = streamPath[:len(streamPath)-4]
|
||||
}
|
||||
|
||||
r.mu.RLock()
|
||||
stream, exists := r.streams[streamPath]
|
||||
r.mu.RUnlock()
|
||||
|
||||
if !exists || !stream.IsActive {
|
||||
http.Error(w, "Stream not found", http.StatusNotFound)
|
||||
return
|
||||
}
|
||||
|
||||
// 设置响应头
|
||||
w.Header().Set("Content-Type", "video/x-flv")
|
||||
w.Header().Set("Access-Control-Allow-Origin", "*")
|
||||
w.Header().Set("Transfer-Encoding", "chunked")
|
||||
|
||||
// TODO: 实际的 FLV 数据推送
|
||||
// 当前为占位实现
|
||||
log.Printf("🎬 [RTMP] FLV 请求: %s", streamPath)
|
||||
|
||||
// 发送 FLV header
|
||||
flvHeader := []byte{0x46, 0x4C, 0x56, 0x01, 0x05, 0x00, 0x00, 0x00, 0x09, 0x00, 0x00, 0x00, 0x00}
|
||||
w.Write(flvHeader)
|
||||
|
||||
// 保持连接直到客户端断开
|
||||
<-req.Context().Done()
|
||||
}
|
||||
|
||||
// handleStreamsAPI 处理流列表 API
|
||||
func (r *RTMPServer) handleStreamsAPI(w http.ResponseWriter, req *http.Request) {
|
||||
w.Header().Set("Content-Type", "application/json")
|
||||
w.Header().Set("Access-Control-Allow-Origin", "*")
|
||||
|
||||
r.mu.RLock()
|
||||
defer r.mu.RUnlock()
|
||||
|
||||
streams := make([]map[string]interface{}, 0)
|
||||
for _, s := range r.streams {
|
||||
if s.IsActive {
|
||||
streams = append(streams, map[string]interface{}{
|
||||
"id": s.ID,
|
||||
"room_id": s.RoomID,
|
||||
"user_id": s.UserID,
|
||||
"pull_url": s.PullURL,
|
||||
"flv_url": s.FLVURL,
|
||||
})
|
||||
}
|
||||
}
|
||||
|
||||
fmt.Fprintf(w, `{"streams":%d,"data":%v}`, len(streams), streams)
|
||||
}
|
||||
|
||||
// Stop 停止 RTMP 服务器
|
||||
func (r *RTMPServer) Stop() {
|
||||
r.mu.Lock()
|
||||
defer r.mu.Unlock()
|
||||
|
||||
r.running = false
|
||||
|
||||
// 关闭 RTMP 监听器
|
||||
if r.rtmpListener != nil {
|
||||
r.rtmpListener.Close()
|
||||
}
|
||||
|
||||
// 关闭 HTTP 服务器
|
||||
if r.httpServer != nil {
|
||||
r.httpServer.Close()
|
||||
}
|
||||
|
||||
// 关闭所有流
|
||||
for _, stream := range r.streams {
|
||||
if stream.stopChan != nil {
|
||||
close(stream.stopChan)
|
||||
}
|
||||
}
|
||||
r.streams = make(map[string]*RTMPStream)
|
||||
|
||||
log.Println("🛑 [RTMP] 已停止")
|
||||
}
|
||||
|
||||
// GenerateStreamURLs 为用户生成推拉流地址
|
||||
func (r *RTMPServer) GenerateStreamURLs(roomID, userID string) (*RTMPStream, error) {
|
||||
r.mu.Lock()
|
||||
defer r.mu.Unlock()
|
||||
|
||||
if !r.running {
|
||||
return nil, ErrRTMPNotReady
|
||||
}
|
||||
|
||||
// 生成唯一的流ID
|
||||
streamID := fmt.Sprintf("%s_%s_%d", roomID, userID, time.Now().UnixNano())
|
||||
|
||||
// 获取服务器配置
|
||||
publicIP := r.config.PublicIP
|
||||
if publicIP == "" {
|
||||
publicIP = viper.GetString("turn.public_ip")
|
||||
}
|
||||
|
||||
// 生成鉴权 Token
|
||||
token := generateStreamToken(streamID)
|
||||
|
||||
// 生成 URL
|
||||
stream := &RTMPStream{
|
||||
ID: streamID,
|
||||
RoomID: roomID,
|
||||
UserID: userID,
|
||||
PushURL: fmt.Sprintf("rtmp://%s:%d/live/%s?token=%s", publicIP, r.config.RTMPPort, streamID, token),
|
||||
PullURL: fmt.Sprintf("rtmp://%s:%d/live/%s", publicIP, r.config.RTMPPort, streamID),
|
||||
FLVURL: fmt.Sprintf("http://%s:%d/live/%s.flv", publicIP, r.config.HTTPFLVPort, streamID),
|
||||
CreatedAt: time.Now(),
|
||||
IsActive: true,
|
||||
dataChan: make(chan []byte, 100),
|
||||
stopChan: make(chan struct{}),
|
||||
}
|
||||
|
||||
r.streams[streamID] = stream
|
||||
log.Printf("🎬 [RTMP] 创建流 | Room:%s User:%s Stream:%s", roomID, userID, streamID)
|
||||
|
||||
return stream, nil
|
||||
}
|
||||
|
||||
// GetStream 获取流信息
|
||||
func (r *RTMPServer) GetStream(streamID string) *RTMPStream {
|
||||
r.mu.RLock()
|
||||
defer r.mu.RUnlock()
|
||||
return r.streams[streamID]
|
||||
}
|
||||
|
||||
// GetStreamsByRoom 获取房间内所有流
|
||||
func (r *RTMPServer) GetStreamsByRoom(roomID string) []*RTMPStream {
|
||||
r.mu.RLock()
|
||||
defer r.mu.RUnlock()
|
||||
|
||||
streams := make([]*RTMPStream, 0)
|
||||
for _, s := range r.streams {
|
||||
if s.RoomID == roomID && s.IsActive {
|
||||
streams = append(streams, s)
|
||||
}
|
||||
}
|
||||
return streams
|
||||
}
|
||||
|
||||
// GetStreamsByUser 获取用户的所有流
|
||||
func (r *RTMPServer) GetStreamsByUser(userID string) []*RTMPStream {
|
||||
r.mu.RLock()
|
||||
defer r.mu.RUnlock()
|
||||
|
||||
streams := make([]*RTMPStream, 0)
|
||||
for _, s := range r.streams {
|
||||
if s.UserID == userID && s.IsActive {
|
||||
streams = append(streams, s)
|
||||
}
|
||||
}
|
||||
return streams
|
||||
}
|
||||
|
||||
// RemoveStream 移除流
|
||||
func (r *RTMPServer) RemoveStream(streamID string) {
|
||||
r.mu.Lock()
|
||||
defer r.mu.Unlock()
|
||||
|
||||
if stream, exists := r.streams[streamID]; exists {
|
||||
stream.IsActive = false
|
||||
if stream.stopChan != nil {
|
||||
close(stream.stopChan)
|
||||
}
|
||||
delete(r.streams, streamID)
|
||||
log.Printf("🎬 [RTMP] 移除流 | Stream:%s", streamID)
|
||||
}
|
||||
}
|
||||
|
||||
// RemoveStreamsByUser 移除用户的所有流
|
||||
func (r *RTMPServer) RemoveStreamsByUser(userID string) {
|
||||
r.mu.Lock()
|
||||
defer r.mu.Unlock()
|
||||
|
||||
for streamID, stream := range r.streams {
|
||||
if stream.UserID == userID {
|
||||
stream.IsActive = false
|
||||
if stream.stopChan != nil {
|
||||
close(stream.stopChan)
|
||||
}
|
||||
delete(r.streams, streamID)
|
||||
log.Printf("🎬 [RTMP] 移除用户流 | User:%s Stream:%s", userID, streamID)
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// RemoveStreamsByRoom 移除房间的所有流
|
||||
func (r *RTMPServer) RemoveStreamsByRoom(roomID string) {
|
||||
r.mu.Lock()
|
||||
defer r.mu.Unlock()
|
||||
|
||||
for streamID, stream := range r.streams {
|
||||
if stream.RoomID == roomID {
|
||||
stream.IsActive = false
|
||||
if stream.stopChan != nil {
|
||||
close(stream.stopChan)
|
||||
}
|
||||
delete(r.streams, streamID)
|
||||
log.Printf("🎬 [RTMP] 移除房间流 | Room:%s Stream:%s", roomID, streamID)
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// GetStreamCount 获取流数量
|
||||
func (r *RTMPServer) GetStreamCount() int {
|
||||
r.mu.RLock()
|
||||
defer r.mu.RUnlock()
|
||||
return len(r.streams)
|
||||
}
|
||||
|
||||
// cleanupTask 清理过期流
|
||||
func (r *RTMPServer) cleanupTask() {
|
||||
ticker := time.NewTicker(5 * time.Minute)
|
||||
defer ticker.Stop()
|
||||
|
||||
for {
|
||||
select {
|
||||
case <-ticker.C:
|
||||
r.cleanupExpiredStreams()
|
||||
}
|
||||
|
||||
r.mu.RLock()
|
||||
if !r.running {
|
||||
r.mu.RUnlock()
|
||||
return
|
||||
}
|
||||
r.mu.RUnlock()
|
||||
}
|
||||
}
|
||||
|
||||
// cleanupExpiredStreams 清理过期的流
|
||||
func (r *RTMPServer) cleanupExpiredStreams() {
|
||||
r.mu.Lock()
|
||||
defer r.mu.Unlock()
|
||||
|
||||
expireTime := time.Now().Add(-1 * time.Hour) // 1小时过期
|
||||
|
||||
for streamID, stream := range r.streams {
|
||||
if stream.CreatedAt.Before(expireTime) && !stream.IsActive {
|
||||
if stream.stopChan != nil {
|
||||
close(stream.stopChan)
|
||||
}
|
||||
delete(r.streams, streamID)
|
||||
log.Printf("🗑️ [RTMP] 清理过期流 | Stream:%s", streamID)
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// generateStreamToken 生成流鉴权 Token
|
||||
func generateStreamToken(streamID string) string {
|
||||
secret := viper.GetString("turn.shared_secret")
|
||||
timestamp := time.Now().Add(24 * time.Hour).Unix()
|
||||
data := fmt.Sprintf("%s:%d", streamID, timestamp)
|
||||
|
||||
mac := hmac.New(sha1.New, []byte(secret))
|
||||
mac.Write([]byte(data))
|
||||
return base64.URLEncoding.EncodeToString(mac.Sum(nil))
|
||||
}
|
||||
|
||||
// ValidateStreamToken 验证流鉴权 Token
|
||||
func ValidateStreamToken(streamID, token string) bool {
|
||||
expectedToken := generateStreamToken(streamID)
|
||||
return token == expectedToken
|
||||
}
|
||||
229
internal/mediaserver/server.go
Normal file
229
internal/mediaserver/server.go
Normal file
@@ -0,0 +1,229 @@
|
||||
/**
|
||||
* package mediaserver
|
||||
*
|
||||
* 媒体服务器主入口
|
||||
* 功能：
|
||||
* 1. 集成 WebRTC SFU（支持 H5/App 端音视频通话）
|
||||
* 2. 集成 RTMP 服务（支持微信小程序 live-pusher/live-player）
|
||||
* 3. 实现 WebRTC ↔ RTMP 媒体桥接（跨平台互通）
|
||||
*/
|
||||
package mediaserver
|
||||
|
||||
import (
|
||||
"log"
|
||||
"sync"
|
||||
|
||||
"github.com/spf13/viper"
|
||||
)
|
||||
|
||||
// MediaServer 媒体服务器主结构
|
||||
type MediaServer struct {
|
||||
sfu *SFUServer // WebRTC SFU 服务
|
||||
rtmp *RTMPServer // RTMP 服务
|
||||
rooms map[string]*Room // 通话房间管理
|
||||
mu sync.RWMutex // 房间锁
|
||||
config *MediaServerConfig // 配置
|
||||
}
|
||||
|
||||
// MediaServerConfig 媒体服务器配置
|
||||
type MediaServerConfig struct {
|
||||
Enabled bool `mapstructure:"enabled"`
|
||||
PublicIP string `mapstructure:"public_ip"`
|
||||
WebRTCPort int `mapstructure:"webrtc_port"`
|
||||
RTMPPort int `mapstructure:"rtmp_port"`
|
||||
HTTPFLVPort int `mapstructure:"http_flv_port"`
|
||||
// SFU 配置
|
||||
MaxRoomSize int `mapstructure:"max_room_size"` // 单房间最大人数
|
||||
// 编解码配置
|
||||
VideoCodec string `mapstructure:"video_codec"` // 视频编解码：h264
|
||||
AudioCodec string `mapstructure:"audio_codec"` // 音频编解码：opus/aac
|
||||
}
|
||||
|
||||
// 全局媒体服务器实例
|
||||
var (
|
||||
server *MediaServer
|
||||
once sync.Once
|
||||
)
|
||||
|
||||
// GetServer 获取媒体服务器单例
|
||||
func GetServer() *MediaServer {
|
||||
once.Do(func() {
|
||||
server = newMediaServer()
|
||||
})
|
||||
return server
|
||||
}
|
||||
|
||||
// newMediaServer 创建媒体服务器实例
|
||||
func newMediaServer() *MediaServer {
|
||||
config := loadConfig()
|
||||
|
||||
ms := &MediaServer{
|
||||
rooms: make(map[string]*Room),
|
||||
config: config,
|
||||
}
|
||||
|
||||
if config.Enabled {
|
||||
ms.sfu = NewSFUServer(config)
|
||||
ms.rtmp = NewRTMPServer(config)
|
||||
}
|
||||
|
||||
return ms
|
||||
}
|
||||
|
||||
// loadConfig 从 viper 加载配置
|
||||
func loadConfig() *MediaServerConfig {
|
||||
config := &MediaServerConfig{
|
||||
Enabled: viper.GetBool("media.enabled"),
|
||||
PublicIP: viper.GetString("media.public_ip"),
|
||||
WebRTCPort: viper.GetInt("media.webrtc_port"),
|
||||
RTMPPort: viper.GetInt("media.rtmp_port"),
|
||||
HTTPFLVPort: viper.GetInt("media.http_flv_port"),
|
||||
MaxRoomSize: viper.GetInt("media.max_room_size"),
|
||||
VideoCodec: viper.GetString("media.video_codec"),
|
||||
AudioCodec: viper.GetString("media.audio_codec"),
|
||||
}
|
||||
|
||||
// 设置默认值
|
||||
if config.PublicIP == "" {
|
||||
config.PublicIP = viper.GetString("turn.public_ip") // 复用 TURN 的公网IP
|
||||
}
|
||||
if config.WebRTCPort == 0 {
|
||||
config.WebRTCPort = 8080
|
||||
}
|
||||
if config.RTMPPort == 0 {
|
||||
config.RTMPPort = 1935
|
||||
}
|
||||
if config.HTTPFLVPort == 0 {
|
||||
config.HTTPFLVPort = 7001
|
||||
}
|
||||
if config.MaxRoomSize == 0 {
|
||||
config.MaxRoomSize = 9 // 默认最多9人
|
||||
}
|
||||
if config.VideoCodec == "" {
|
||||
config.VideoCodec = "h264"
|
||||
}
|
||||
if config.AudioCodec == "" {
|
||||
config.AudioCodec = "opus"
|
||||
}
|
||||
|
||||
return config
|
||||
}
|
||||
|
||||
// Start 启动媒体服务器
|
||||
func Start() {
|
||||
ms := GetServer()
|
||||
|
||||
if !ms.config.Enabled {
|
||||
log.Println("ℹ️ [MediaServer] 服务已禁用")
|
||||
return
|
||||
}
|
||||
|
||||
log.Println("🎥 [MediaServer] 正在启动...")
|
||||
|
||||
// 启动 WebRTC SFU
|
||||
if ms.sfu != nil {
|
||||
go ms.sfu.Start()
|
||||
}
|
||||
|
||||
// 启动 RTMP 服务
|
||||
if ms.rtmp != nil {
|
||||
go ms.rtmp.Start()
|
||||
}
|
||||
|
||||
log.Printf("🎥 [MediaServer] 已启动 | WebRTC: %d | RTMP: %d | HTTP-FLV: %d",
|
||||
ms.config.WebRTCPort, ms.config.RTMPPort, ms.config.HTTPFLVPort)
|
||||
}
|
||||
|
||||
// Stop 停止媒体服务器
|
||||
func (ms *MediaServer) Stop() {
|
||||
log.Println("🛑 [MediaServer] 正在停止...")
|
||||
|
||||
// 关闭所有房间
|
||||
ms.mu.Lock()
|
||||
for _, room := range ms.rooms {
|
||||
room.Close()
|
||||
}
|
||||
ms.rooms = make(map[string]*Room)
|
||||
ms.mu.Unlock()
|
||||
|
||||
// 停止 SFU
|
||||
if ms.sfu != nil {
|
||||
ms.sfu.Stop()
|
||||
}
|
||||
|
||||
// 停止 RTMP
|
||||
if ms.rtmp != nil {
|
||||
ms.rtmp.Stop()
|
||||
}
|
||||
|
||||
log.Println("🛑 [MediaServer] 已停止")
|
||||
}
|
||||
|
||||
// GetOrCreateRoom 获取或创建房间
|
||||
func (ms *MediaServer) GetOrCreateRoom(roomID string) *Room {
|
||||
ms.mu.Lock()
|
||||
defer ms.mu.Unlock()
|
||||
|
||||
if room, exists := ms.rooms[roomID]; exists {
|
||||
return room
|
||||
}
|
||||
|
||||
room := NewRoom(roomID, ms.config.MaxRoomSize)
|
||||
ms.rooms[roomID] = room
|
||||
log.Printf("🎥 [MediaServer] 创建房间: %s", roomID)
|
||||
return room
|
||||
}
|
||||
|
||||
// GetRoom 获取房间
|
||||
func (ms *MediaServer) GetRoom(roomID string) *Room {
|
||||
ms.mu.RLock()
|
||||
defer ms.mu.RUnlock()
|
||||
return ms.rooms[roomID]
|
||||
}
|
||||
|
||||
// RemoveRoom 移除房间
|
||||
func (ms *MediaServer) RemoveRoom(roomID string) {
|
||||
ms.mu.Lock()
|
||||
defer ms.mu.Unlock()
|
||||
|
||||
if room, exists := ms.rooms[roomID]; exists {
|
||||
room.Close()
|
||||
delete(ms.rooms, roomID)
|
||||
log.Printf("🎥 [MediaServer] 移除房间: %s", roomID)
|
||||
}
|
||||
}
|
||||
|
||||
// GetConfig 获取配置
|
||||
func (ms *MediaServer) GetConfig() *MediaServerConfig {
|
||||
return ms.config
|
||||
}
|
||||
|
||||
// GetSFU 获取 SFU 服务
|
||||
func (ms *MediaServer) GetSFU() *SFUServer {
|
||||
return ms.sfu
|
||||
}
|
||||
|
||||
// GetRTMP 获取 RTMP 服务
|
||||
func (ms *MediaServer) GetRTMP() *RTMPServer {
|
||||
return ms.rtmp
|
||||
}
|
||||
|
||||
// GetRoomCount 获取房间数量
|
||||
func (ms *MediaServer) GetRoomCount() int {
|
||||
ms.mu.RLock()
|
||||
defer ms.mu.RUnlock()
|
||||
return len(ms.rooms)
|
||||
}
|
||||
|
||||
// GetAllRooms 获取所有房间信息（用于调试/监控）
|
||||
func (ms *MediaServer) GetAllRooms() []*RoomInfo {
|
||||
ms.mu.RLock()
|
||||
defer ms.mu.RUnlock()
|
||||
|
||||
infos := make([]*RoomInfo, 0, len(ms.rooms))
|
||||
for _, room := range ms.rooms {
|
||||
infos = append(infos, room.GetInfo())
|
||||
}
|
||||
return infos
|
||||
}
|
||||
|
||||
395
internal/mediaserver/sfu.go
Normal file
395
internal/mediaserver/sfu.go
Normal file
@@ -0,0 +1,395 @@
|
||||
/**
|
||||
* package mediaserver
|
||||
*
|
||||
* WebRTC SFU (Selective Forwarding Unit) 服务
|
||||
* 功能：
|
||||
* 1. 接收 WebRTC 流
|
||||
* 2. 转发给同房间的其他用户
|
||||
* 3. 支持音视频分离转发
|
||||
*/
|
||||
package mediaserver
|
||||
|
||||
import (
|
||||
"crypto/hmac"
|
||||
"crypto/sha1"
|
||||
"encoding/base64"
|
||||
"encoding/json"
|
||||
"fmt"
|
||||
"log"
|
||||
"strings"
|
||||
"sync"
|
||||
"time"
|
||||
|
||||
"github.com/pion/interceptor"
|
||||
"github.com/pion/interceptor/pkg/intervalpli"
|
||||
"github.com/pion/webrtc/v3"
|
||||
"github.com/spf13/viper"
|
||||
)
|
||||
|
||||
// SFUServer WebRTC SFU 服务器
|
||||
type SFUServer struct {
|
||||
config *MediaServerConfig
|
||||
api *webrtc.API
|
||||
mu sync.RWMutex
|
||||
running bool
|
||||
|
||||
// Track 管理
|
||||
trackLocals map[string]*webrtc.TrackLocalStaticRTP // trackID -> localTrack
|
||||
}
|
||||
|
||||
// NewSFUServer 创建 SFU 服务器
|
||||
func NewSFUServer(config *MediaServerConfig) *SFUServer {
|
||||
return &SFUServer{
|
||||
config: config,
|
||||
trackLocals: make(map[string]*webrtc.TrackLocalStaticRTP),
|
||||
}
|
||||
}
|
||||
|
||||
// Start 启动 SFU 服务器
|
||||
func (s *SFUServer) Start() {
|
||||
s.mu.Lock()
|
||||
if s.running {
|
||||
s.mu.Unlock()
|
||||
return
|
||||
}
|
||||
s.running = true
|
||||
s.mu.Unlock()
|
||||
|
||||
// 创建 MediaEngine
|
||||
m := &webrtc.MediaEngine{}
|
||||
|
||||
// 注册默认编解码器
|
||||
if err := m.RegisterDefaultCodecs(); err != nil {
|
||||
log.Printf("❌ [SFU] 注册编解码器失败: %v", err)
|
||||
return
|
||||
}
|
||||
|
||||
// 创建拦截器注册表
|
||||
i := &interceptor.Registry{}
|
||||
|
||||
// 注册 PLI 拦截器（用于请求关键帧）
|
||||
intervalPliFactory, err := intervalpli.NewReceiverInterceptor()
|
||||
if err != nil {
|
||||
log.Printf("❌ [SFU] 创建 PLI 拦截器失败: %v", err)
|
||||
return
|
||||
}
|
||||
i.Add(intervalPliFactory)
|
||||
|
||||
// 使用拦截器
|
||||
if err := webrtc.RegisterDefaultInterceptors(m, i); err != nil {
|
||||
log.Printf("❌ [SFU] 注册拦截器失败: %v", err)
|
||||
return
|
||||
}
|
||||
|
||||
// 创建 API
|
||||
s.api = webrtc.NewAPI(webrtc.WithMediaEngine(m), webrtc.WithInterceptorRegistry(i))
|
||||
|
||||
log.Printf("🚀 [SFU] WebRTC SFU 已启动 | 端口: %d", s.config.WebRTCPort)
|
||||
}
|
||||
|
||||
// Stop 停止 SFU 服务器
|
||||
func (s *SFUServer) Stop() {
|
||||
s.mu.Lock()
|
||||
defer s.mu.Unlock()
|
||||
|
||||
s.running = false
|
||||
s.trackLocals = make(map[string]*webrtc.TrackLocalStaticRTP)
|
||||
log.Println("🛑 [SFU] 已停止")
|
||||
}
|
||||
|
||||
// CreatePeerConnection 为用户创建 PeerConnection
|
||||
func (s *SFUServer) CreatePeerConnection(roomID, userID string) (*webrtc.PeerConnection, error) {
|
||||
s.mu.RLock()
|
||||
if !s.running || s.api == nil {
|
||||
s.mu.RUnlock()
|
||||
return nil, ErrSFUNotReady
|
||||
}
|
||||
s.mu.RUnlock()
|
||||
|
||||
// 获取 ICE 服务器配置
|
||||
iceServers := s.getICEServers(userID)
|
||||
|
||||
// 创建 PeerConnection 配置
|
||||
config := webrtc.Configuration{
|
||||
ICEServers: iceServers,
|
||||
}
|
||||
|
||||
// 创建 PeerConnection
|
||||
pc, err := s.api.NewPeerConnection(config)
|
||||
if err != nil {
|
||||
return nil, fmt.Errorf("create peer connection failed: %w", err)
|
||||
}
|
||||
|
||||
// 监听 ICE 连接状态
|
||||
pc.OnICEConnectionStateChange(func(state webrtc.ICEConnectionState) {
|
||||
log.Printf("🔗 [SFU] Room:%s User:%s ICE状态: %s", roomID, userID, state.String())
|
||||
|
||||
if state == webrtc.ICEConnectionStateFailed || state == webrtc.ICEConnectionStateDisconnected {
|
||||
// 连接断开，清理资源
|
||||
s.handleDisconnect(roomID, userID)
|
||||
}
|
||||
})
|
||||
|
||||
// 监听轨道
|
||||
pc.OnTrack(func(remoteTrack *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {
|
||||
s.handleTrack(roomID, userID, remoteTrack, receiver)
|
||||
})
|
||||
|
||||
return pc, nil
|
||||
}
|
||||
|
||||
// getICEServers 获取 ICE 服务器配置
|
||||
func (s *SFUServer) getICEServers(userID string) []webrtc.ICEServer {
|
||||
servers := []webrtc.ICEServer{}
|
||||
|
||||
// 添加 STUN 服务器
|
||||
turnPublicIP := viper.GetString("turn.public_ip")
|
||||
turnPort := viper.GetInt("turn.listen_port")
|
||||
if turnPublicIP != "" && turnPort > 0 {
|
||||
stunURL := fmt.Sprintf("stun:%s:%d", turnPublicIP, turnPort)
|
||||
servers = append(servers, webrtc.ICEServer{
|
||||
URLs: []string{stunURL},
|
||||
})
|
||||
|
||||
// 添加 TURN 服务器（如果启用）
|
||||
if viper.GetBool("turn.enabled") {
|
||||
turnURL := fmt.Sprintf("turn:%s:%d", turnPublicIP, turnPort)
|
||||
// 生成 TURN 凭证
|
||||
username, credential := generateTURNCredentials(userID)
|
||||
servers = append(servers, webrtc.ICEServer{
|
||||
URLs: []string{turnURL},
|
||||
Username: username,
|
||||
Credential: credential,
|
||||
})
|
||||
}
|
||||
}
|
||||
|
||||
return servers
|
||||
}
|
||||
|
||||
// handleTrack 处理远程轨道
|
||||
func (s *SFUServer) handleTrack(roomID, userID string, remoteTrack *webrtc.TrackRemote, receiver *webrtc.RTPReceiver) {
|
||||
trackID := fmt.Sprintf("%s_%s_%s", roomID, userID, remoteTrack.Kind().String())
|
||||
|
||||
log.Printf("🎬 [SFU] 收到轨道 | Room:%s User:%s Kind:%s ID:%s",
|
||||
roomID, userID, remoteTrack.Kind().String(), trackID)
|
||||
|
||||
// 创建本地轨道用于转发
|
||||
localTrack, err := webrtc.NewTrackLocalStaticRTP(
|
||||
remoteTrack.Codec().RTPCodecCapability,
|
||||
trackID,
|
||||
fmt.Sprintf("stream_%s", userID),
|
||||
)
|
||||
if err != nil {
|
||||
log.Printf("❌ [SFU] 创建本地轨道失败: %v", err)
|
||||
return
|
||||
}
|
||||
|
||||
// 保存轨道
|
||||
s.mu.Lock()
|
||||
s.trackLocals[trackID] = localTrack
|
||||
s.mu.Unlock()
|
||||
|
||||
// 转发 RTP 包
|
||||
go func() {
|
||||
buf := make([]byte, 1500)
|
||||
for {
|
||||
n, _, readErr := remoteTrack.Read(buf)
|
||||
if readErr != nil {
|
||||
log.Printf("⚠️ [SFU] 读取轨道失败 | Track:%s Error:%v", trackID, readErr)
|
||||
break
|
||||
}
|
||||
|
||||
// 写入本地轨道（会自动转发给所有订阅者）
|
||||
if _, writeErr := localTrack.Write(buf[:n]); writeErr != nil {
|
||||
log.Printf("⚠️ [SFU] 写入轨道失败 | Track:%s Error:%v", trackID, writeErr)
|
||||
break
|
||||
}
|
||||
}
|
||||
|
||||
// 清理轨道
|
||||
s.mu.Lock()
|
||||
delete(s.trackLocals, trackID)
|
||||
s.mu.Unlock()
|
||||
}()
|
||||
|
||||
// 通知房间内其他用户有新轨道
|
||||
s.notifyNewTrack(roomID, userID, localTrack)
|
||||
}
|
||||
|
||||
// notifyNewTrack 通知房间内其他用户有新轨道
|
||||
func (s *SFUServer) notifyNewTrack(roomID, senderUserID string, track *webrtc.TrackLocalStaticRTP) {
|
||||
ms := GetServer()
|
||||
room := ms.GetRoom(roomID)
|
||||
if room == nil {
|
||||
return
|
||||
}
|
||||
|
||||
// 获取其他参与者
|
||||
for _, p := range room.GetOtherParticipants(senderUserID) {
|
||||
if p.Type != ParticipantTypeWebRTC {
|
||||
continue // 只处理 WebRTC 用户
|
||||
}
|
||||
|
||||
if p.PeerConnection == nil {
|
||||
continue
|
||||
}
|
||||
|
||||
pc, ok := p.PeerConnection.(*webrtc.PeerConnection)
|
||||
if !ok {
|
||||
continue
|
||||
}
|
||||
|
||||
// 添加轨道到对方的 PeerConnection
|
||||
if _, err := pc.AddTrack(track); err != nil {
|
||||
log.Printf("⚠️ [SFU] 添加轨道到用户 %s 失败: %v", p.UserID, err)
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// handleDisconnect 处理断开连接
|
||||
func (s *SFUServer) handleDisconnect(roomID, userID string) {
|
||||
ms := GetServer()
|
||||
room := ms.GetRoom(roomID)
|
||||
if room == nil {
|
||||
return
|
||||
}
|
||||
|
||||
room.RemoveParticipant(userID)
|
||||
|
||||
// 清理该用户的所有轨道
|
||||
s.mu.Lock()
|
||||
for trackID := range s.trackLocals {
|
||||
// 检查 trackID 是否属于该用户
|
||||
prefix := fmt.Sprintf("%s_%s_", roomID, userID)
|
||||
if strings.HasPrefix(trackID, prefix) {
|
||||
delete(s.trackLocals, trackID)
|
||||
}
|
||||
}
|
||||
s.mu.Unlock()
|
||||
|
||||
// 如果房间空了，移除房间
|
||||
if room.IsEmpty() {
|
||||
ms.RemoveRoom(roomID)
|
||||
}
|
||||
}
|
||||
|
||||
// GetTrackCount 获取轨道数量
|
||||
func (s *SFUServer) GetTrackCount() int {
|
||||
s.mu.RLock()
|
||||
defer s.mu.RUnlock()
|
||||
return len(s.trackLocals)
|
||||
}
|
||||
|
||||
// HandleOffer 处理 SDP Offer
|
||||
func (s *SFUServer) HandleOffer(roomID, userID string, offerSDP string) (string, error) {
|
||||
ms := GetServer()
|
||||
room := ms.GetOrCreateRoom(roomID)
|
||||
|
||||
// 添加参与者
|
||||
participant, err := room.AddParticipant(userID, ParticipantTypeWebRTC)
|
||||
if err != nil {
|
||||
return "", err
|
||||
}
|
||||
|
||||
// 创建 PeerConnection
|
||||
pc, err := s.CreatePeerConnection(roomID, userID)
|
||||
if err != nil {
|
||||
return "", err
|
||||
}
|
||||
participant.PeerConnection = pc
|
||||
|
||||
// 设置远程描述
|
||||
offer := webrtc.SessionDescription{
|
||||
Type: webrtc.SDPTypeOffer,
|
||||
SDP: offerSDP,
|
||||
}
|
||||
if err := pc.SetRemoteDescription(offer); err != nil {
|
||||
return "", fmt.Errorf("set remote description failed: %w", err)
|
||||
}
|
||||
|
||||
// 添加房间内其他用户的轨道
|
||||
s.addExistingTracks(pc, roomID, userID)
|
||||
|
||||
// 创建 Answer
|
||||
answer, err := pc.CreateAnswer(nil)
|
||||
if err != nil {
|
||||
return "", fmt.Errorf("create answer failed: %w", err)
|
||||
}
|
||||
|
||||
// 设置本地描述
|
||||
if err := pc.SetLocalDescription(answer); err != nil {
|
||||
return "", fmt.Errorf("set local description failed: %w", err)
|
||||
}
|
||||
|
||||
// 等待 ICE 收集完成
|
||||
<-webrtc.GatheringCompletePromise(pc)
|
||||
|
||||
return pc.LocalDescription().SDP, nil
|
||||
}
|
||||
|
||||
// addExistingTracks 添加房间内已有的轨道
|
||||
func (s *SFUServer) addExistingTracks(pc *webrtc.PeerConnection, roomID, excludeUserID string) {
|
||||
s.mu.RLock()
|
||||
defer s.mu.RUnlock()
|
||||
|
||||
prefix := fmt.Sprintf("%s_", roomID)
|
||||
excludePrefix := fmt.Sprintf("%s_%s_", roomID, excludeUserID)
|
||||
|
||||
for trackID, track := range s.trackLocals {
|
||||
// 检查是否是同房间的轨道，且不是自己的
|
||||
if strings.HasPrefix(trackID, prefix) && !strings.HasPrefix(trackID, excludePrefix) {
|
||||
if _, err := pc.AddTrack(track); err != nil {
|
||||
log.Printf("⚠️ [SFU] 添加已有轨道失败: %v", err)
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
// HandleICECandidate 处理 ICE 候选
|
||||
func (s *SFUServer) HandleICECandidate(roomID, userID string, candidateJSON string) error {
|
||||
ms := GetServer()
|
||||
room := ms.GetRoom(roomID)
|
||||
if room == nil {
|
||||
return ErrRoomNotFound
|
||||
}
|
||||
|
||||
participant := room.GetParticipant(userID)
|
||||
if participant == nil {
|
||||
return ErrParticipantNotFound
|
||||
}
|
||||
|
||||
pc, ok := participant.PeerConnection.(*webrtc.PeerConnection)
|
||||
if !ok || pc == nil {
|
||||
return ErrSFUNotReady
|
||||
}
|
||||
|
||||
var candidate webrtc.ICECandidateInit
|
||||
if err := json.Unmarshal([]byte(candidateJSON), &candidate); err != nil {
|
||||
return fmt.Errorf("parse ICE candidate failed: %w", err)
|
||||
}
|
||||
|
||||
if err := pc.AddICECandidate(candidate); err != nil {
|
||||
return fmt.Errorf("add ICE candidate failed: %w", err)
|
||||
}
|
||||
|
||||
return nil
|
||||
}
|
||||
|
||||
// generateTURNCredentials 生成 TURN 凭证
|
||||
func generateTURNCredentials(userID string) (string, string) {
|
||||
timestamp := time.Now().Add(24 * time.Hour).Unix()
|
||||
username := fmt.Sprintf("%d:%s", timestamp, userID)
|
||||
secret := viper.GetString("turn.shared_secret")
|
||||
|
||||
mac := hmac.New(sha1.New, []byte(secret))
|
||||
mac.Write([]byte(username))
|
||||
password := base64.StdEncoding.EncodeToString(mac.Sum(nil))
|
||||
|
||||
return username, password
|
||||
}
|
||||
|
||||
// GetTURNCredentials 获取 TURN 凭证（供外部调用）
|
||||
func GetTURNCredentials(userID string) (string, string) {
|
||||
return generateTURNCredentials(userID)
|
||||
}
|
||||
Reference in New Issue
Block a user