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Notes for my learning.

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Webrtc

jinleileiking opened this issue · comments

commented

建链流程

sequenceDiagram
    par 
    Server-->>Browser: sending offer
    Browser->>Browser: subpc.setRemoteDescription
    and

    and
    Server-->>Browser: sending trickle * n
    Browser->>Browser: add to queue
    end
    Browser->>Browser: subpc.addIceCandidate * n
    Browser->>Browser: subpc.setLocalDescription
    Browser-->>Server: sending answer

    rect rgb(191, 223, 255)
    Browser->>Browser: pubpc.onconnectionstatechange(connecting)
    Browser->>Browser: subpc.onicecandidate
    Browser-->>Server: sending trickle
    end

    rect rgb(191, 223, 255)
    Browser->>Browser: pubpc.onconnectionstatechange(connected)
    end
Loading

sdp

o=sessionId sessionVersion ipCreatedSDP
a=group:BUNDLE  --- establishes a relationship between several media lines included in the SDP
a=msid-semantic WMS
m= meida line
c= ---- my ip
a=rtcp --- rtcp uses port
a=candidate:    1/2 [rtp/rtcp] udp  [priority] 
a=extmap

不懂的sdp

a=fmtp:100 profile-id=2
a=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=42001f

https://blog.csdn.net/china_jeffery/article/details/79991986

Rtcp信息

Chrome作为发送方，会发送:

  • 500 ms 发送一个 Sender Report

Chrome作为接收方：

  • RR (201) 2s一个
  • AVB RTCP (208) 和RR 一起发送
  • Summary (209) 和RR 一起发送
  • Generic-FeedBack (205)
    • NACK
  • Payload-Feedback (206)
    • PLI

svc

Sender

  • frag
  • gen timestamp
  • gen error correction
  • reorder frames
  • ssrs colision
  • congrestion controll

Congestion Control

  • TCP-Friendly Rate Control
  • RTSP (RTP over TCP)
  • RAP (Rate Adaptation Protocol)

Other

  • CRTP
  • ROHC

dtls

pion作为client的cipher suite：

c0 2b Cipher Suite: TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 (0xc02b)
c0 2f Cipher Suite: TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256 (0xc02f)
c0 0a Cipher Suite: TLS_ECDHE_ECDSA_WITH_AES_256_CBC_SHA (0xc00a)
00 35 Cipher Suite: TLS_RSA_WITH_AES_256_CBC_SHA (0x0035)

Arch

  • signal server collects basic info from sfus stores to redis
    • cpu
    • streams count
    • ....
  • client1 (the first to join room) sends request to signal server, return sdps and which sfu to connect, save the infos to redis
  • client2 (not the first to join room) sends request to signal server, return sdps and which sfu to connect ( client1 sfu ), save the infos to redis

References

Book

  • RTP: Audio and Video for the Internet
  • Learning WebRTC
  • Getting Started with WebRTC
  • Real-Time Communication with WebRTC
  • webrtc cookbook

RFCs

Proposal

Standards

Urls

Codes