Webrtc

[jinleileiking]

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

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

• sdp: https://www.rfc-editor.org/rfc/rfc6190#section-7.3.1

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

• https://w3c.github.io/webrtc-quic/

Standards

• https://tools.ietf.org/wg/rtcweb/
• https://webrtc.org/
• https://www.w3.org/TR/webrtc/

Urls

• https://developer.ridgerun.com/wiki/index.php?title=GstWebRTC_-_WebRTC_Fundamentals
• https://hpbn.co/webrtc/
• https://github.com/pions/webrtc/wiki/Relative-Drafts-And-Usages

Codes

• https://github.com/pions/webrtc
• https://github.com/spebsd/live-webrtcsignaling

[jinleileiking]

TWCC：

https://blog.csdn.net/Birdyxh/article/details/130249884?spm=1001.2101.3001.6650.2&utm_medium=distribute.pc_relevant.none-task-blog-2%7Edefault%7EYuanLiJiHua%7EPosition-2-130249884-blog-123405649.235%5Ev38%5Epc_relevant_sort_base3&depth_1-utm_source=distribute.pc_relevant.none-task-blog-2%7Edefault%7EYuanLiJiHua%7EPosition-2-130249884-blog-123405649.235%5Ev38%5Epc_relevant_sort_base3&utm_relevant_index=5

https://blog.jianchihu.net/webrtc-research-transport-cc-rtp-rtcp.html