---
title: Lab Meeting Minutes 2022/03/25
tags: lab_meeting
---
> Outline
> [TOC]
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# PERAL Lab Meeting
- 時間：111 年 03 月 25 日 10:00
- 地點：TC 321
- 出席者：吳坤熹老師、謝萬霖、周以恆、吳騰然、劉怡君、田蕙瑜、洪胤勛、丘世宇、莊才賢

- 會議主題： [WebRTC](http://ms15.voip.edu.tw/~webmaster/meeting/1102/PPT/webRTC_v2.pptx)
- 主講者: 莊才賢
- 主記: 謝萬霖

## 會議內容
- WebRTC feature
- Acquire audio and video
- Communicating data
- History
- Licenses are used to be expensive
- RTC is time-comsuming on web
- Google IP Solution (GIPS) is open-sourced
- a great push on WebRTC
- Session Description Protocol (SDP)
- RFC 8866
- Describe session, not transport nor content
- Can work with
- HTTP
- SIP
- SAP
- RTSP
- SDP format (Session description)
- Protocol version (v=)
- default 0
- Origin (o=)
- `<username> <sess-id> <sess-version> <nettype> <addrtype> <unicast-address>`
- `username` if not support, `"-"`
- Connection Information (c=)
- `<nettype> <addrtype> <connection-address>`
- if no candidate, insert dummy
- e.g. `c=IN IP4 0.0.0.0`
- Session Name (s=)
- `<session name>`
- if no meaningful, use `s="-"`
- Bandwidth Information (b=)
- Session Information (i=)
- URI (u=)`
- Email Address (e=)
- Phone Number (p=)
- SDP format (Time description)
- Not important
- SDP format (Media description)
- Media Description (m=)
- start from "m=", end at next "m=" and end of session
- SDP can contain more than one "m="
- The prior the higher priority
- `<media> <port> <proto> <fmt> ...`
- Offer/Answer model
- Purpose
- Initialize session
- Update session
- Initialize the session
1. Offerer send SDP offer to Answerer
- Describe local environment
3. Answerer reply answer to offerer
- describe the overlap of local environment and offerer’s encoding and codec
- Update session
1. new SDP offer/answer can achieve this behavior
2. the offer’s modification can be reject by anser at any reason
- Interactive Connectivity Establishment (ICE)
- find a best route systematically to communicate over the NAT or firewall
- a systematically methods to find best route
- Steps
1. gather all canidates address
2. prioritizing candidates
3. eliminating redundant candidates
4. connectivity check
5. sort canidate
6. find the best route
- WebRTC APIs
- MediaStream (aka getUserMedia)
- RTCPeerConnection
- RTCDataChannel
- MediaStream
- get the stream from browser
- may contain multi tracks
- RTCPeerConnection
- Signal processing
- Codec handling
- P2P communcation
- Security
- Bandwitdth management
- browser 提供簡單 api 簡化建立 p2p connection 過程
- 除了提供協商通道和傳輸資料，剩下的操作被包裝在 browser
- Establish RTCPeerConnection
1. signaling
2. constraints
3. peer connection
4. Run callbacks
- onnegotiationneeded
- 設定 local 的 sdp offer
- onicecandidate
- 送出 local 可行網路的 sdp
- onTrack
- 設定傳回的 track 用途
- onMessage
- 接收 sdp offer 儲存到 peer connection 並產生 sdp answer 回傳給傳遞 offer 的 user
- 把 candidate 加入到 peer connection 讓瀏覽器底層進行協商
- Image explanation
- webRTC’s offer/answer model
- ICE’s behavior
- Demo
- Architecture

### 建議&問題
1. Music on Hold 傳送音樂過去，怎樣叫效能好？[name=Solomon]
Ans : 1. 送整段音樂過去 2. 送 Signal 過去播 local 音樂 [name=Solomon]
2. 可以在 history 之後先簡單介紹 WebRTC 模組，這樣在說明 SDP 和 ICE 才比較不容易迷失。[name=Edgar]

## 待追蹤事項
1. [name=]
Ans : [name=]

## 臨時動議
1. [name=]
Ans : [name=]

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散會結束時間： 12:00