109 lines
4.2 KiB
Markdown
109 lines
4.2 KiB
Markdown
# 媒体
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只要负责媒体处理，不要添加任何业务逻辑，所有业务逻辑都由[taoyao-signal-server](../taoyao-signal-server)处理。
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## 使用
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```
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sudo npm install
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```
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## Mediasoup
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```
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# 编译：默认不用手动编译
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make
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make -C worker
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```
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## 事件
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```
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mediasoup.observer.on("newworker", fn(worker));
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worker.on("died", fn(error));
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worker.observer.on("close", fn());
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worker.observer.on("newrouter", fn(router));
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worker.observer.on("newwebrtcserver", fn(router));
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router.on(“workerclose”, fn());
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router.observer.on(“close”, fn());
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router.observer.on(“newtransport”, fn(transport));
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router.observer.on(“newrtpobserver”, fn(rtpObserver));
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transport.on("trace", fn(trace));
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transport.on(“routerclose”, fn());
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transport.on(“listenserverclose”, fn());
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transport.on(“trace”, fn(trace));
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webRtcTransport.on(“icestatechange”, fn(iceState))
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webRtcTransport.on(“iceselectedtuplechange”, fn(iceSelectedTuple))
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webRtcTransport.on(“dtlsstatechange”, fn(dtlsState))
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webRtcTransport.on(“sctpstatechange”, fn(sctpState))
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plainTransport.on(“tuple”, fn(tuple))
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plainTransport.on(“rtcptuple”, fn(rtcpTuple))
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plainTransport.on(“sctpstatechange”, fn(sctpState))
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pipeTransport.on(“sctpstatechange”, fn(sctpState))
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directTransport.on(“rtcp”, fn(rtcpPacket))
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transport.observer.on(“close”, fn())
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transport.observer.on(“newproducer”, fn(producer))
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transport.observer.on(“newconsumer”, fn(consumer))
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transport.observer.on(“newdataproducer”, fn(dataProducer))
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transport.observer.on(“newdataconsumer”, fn(dataConsumer))
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transport.observer.on(“trace”, fn(trace))
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webRtcTransport.observer.on(“icestatechange”, fn(iceState))
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webRtcTransport.observer.on(“iceselectedtuplechange”, fn(iceSelectedTuple))
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webRtcTransport.observer.on(“dtlsstatechange”, fn(dtlsState))
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webRtcTransport.observer.on(“sctpstatechange”, fn(sctpState))
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plainTransport.observer.on(“tuple”, fn(tuple))
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plainTransport.observer.on(“rtcptuple”, fn(rtcpTuple))
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plainTransport.observer.on(“sctpstatechange”, fn(sctpState))
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pipeTransport.observer.on(“sctpstatechange”, fn(sctpState))
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```
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## 安全
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默认媒体服务只要暴露媒体`UDP`端口，信令接口不用暴露，所以使用简单鉴权。
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## 动态调节码率
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```
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参考配置`mediaCodecs`
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```
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## WebRTC协议栈
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```
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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| HTTPS / WSS | | SCTP | SRTP / SRTCP |
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ ICE / SDP / SIP +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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| TLS | | |
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ DTLS +-+-+-+-+-+-+-+-+-+
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| HTTP / WS | NAT / STUN / TURN | | RTP / RTCP |
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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| TCP | UDP |
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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| IPv4 / IPv6 |
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+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
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```
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### 协议简介
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* 会话通道：ICE/SIP/SDP
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* 媒体通道：RTP/RTCP/SRTP/SRTCP
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* RTP：实时传输协议（音频视频）
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* RTCP：RTP传输控制协议（监控数据传输质量并给予数据发送方反馈）
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* SCTP：流控制传输协议（自定义的应用数据传输）
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* RTMP：实时消息传送协议
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* RTSP：可以控制媒体（点播）
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### ICE/SDP/SIP
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ICE信息的描述格式通常采用标准的SDP，其全称为Session Description Protocol，即会话描述协议。<br />
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SDP只是一种信息格式的描述标准，不属于传输协议，但是可以被其他传输协议用来交换必要的信息，例如：SIP、RTSP等等。
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## 其他常见WebRTC媒体服务
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* [Jitsi](https://github.com/jitsi)
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* [Janus](https://github.com/meetecho/janus-gateway/)
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* [Licode](https://github.com/lynckia/licode)
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* [Kurento](https://github.com/Kurento/kurento-media-server)
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* [Medooze](https://github.com/medooze/media-server)
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* [Mediasoup](https://github.com/versatica/mediasoup)
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