changjurhee / aone Goto Github PK
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License: MIT License
2023 LG SA
License: MIT License
Webtrc 코드 적용 검토를 진행 중입니다.
현재 gstreamer에서 암호화 element에 대한 적용 검토를 하고 있습니다.
2>C:\Program Files\Microsoft Visual Studio\2022\Professional\MSBuild\Microsoft\VC\v170\Microsoft.CppCommon.targets(2387,5): error MSB8011: 출력을 등록하지 못했습니다. 사용자 단위 리디렉션을 사용하도록 설정하거나 높은 권한으로 명령 프롬프트에서 구성 요소를 등록하십시오.
N | Category | Description | responsibility |
---|---|---|---|
1 | User account | System shall allow user to register account with email and password If there is no user who has already registered with the same email | session |
2 | User account | System shall allow user to change user information. | session |
3 | User account | System shall provide user to log in using the previously registered email and password and server ID address in order to acquire the permission to use the system | session |
4 | User account | System shall allow administrator to view / enable / disable / delete user accounts | session |
5 | User account | System shall allow user to create own contact list by searching other registered user accounts using Email or Name | session |
6 | User account | System shall allow user to add / remove contact item from own contact list. | session |
7 | User account | System shall save user password in secure way not to expored by attackers who extract user data from server | |
8 | User Interface | System shall brought application to the foreground when incoming call alerts while application is minimized or another app is in foreground | session |
9 | Direct call | System shall allow user to initiate direct call using previously created contacts list | session/media |
10 | Direct call | System shall allow user to cancel direct call during call initiation | session/media |
11 | Direct call | System shall allow user to end direct call while direct call is in active state | session/media |
12 | Direct call | System shall show user outgoing status after call initiation until receiving user takes an action | session/media |
13 | Direct call | System shall show user rejected information from remote user during direct call initiation | session/media |
14 | Direct call | System shall show user state informations (User Busy, Dropped, Network error) during direct call initiation. | |
15 | Direct call | System shall show user incoming direct call notification during idle status. | session/media |
16 | Direct call | System shall allow user to answer or reject incoming call via incoming direct call notification | session/media |
17 | Conference call | System shall allow user to register(schedule) Conference call with participant's information and date, time, and duration. | session/media |
18 | Conference call | System shall notify all participants of the conference call about the conference started when the registered conference start time of conference arrived | session/media |
19 | Conference call | System shall allow user to join conference sessions that include users when the current time is the same as the Conference's time | session/media |
20 | Conference call | System shall allow user to quit conference session during active conference call status | session/media |
21 | Call session | System shall show video of every participants during active call / conference status | session/media |
22 | Call session | System shall show contact information of every participants during active call / conference status | session/media |
23 | Call session | System shall play audio of every participaints during active call / conference status | session/media |
24 | Call session | System shall provide user acoustic echo cancellation (AEC) during active call / conference status | session/media |
25 | Call session | System shall provide user voice activation detection (VAD) during active call / conference status | session/media |
26 | Call session | System shall provides a graphical highlight to identify which participant is currently speaking during active call / conference status | session/media |
27 | Call session | System shall transfer data for sessions, video data, audio streaming data between server & client in secure way. | session/media |
How to measure the latency audio and video between clients
#if 0 // 지웅님 아래 내용 참고 하세요.
ServerMediaManager* test = ServerMediaManager::getInstance();
CVoIPClientView* pView = (CVoIPClientView*)((CMainFrame*)AfxGetMainWnd())->GetActiveView();
test->setViewHandler((handleptr)pView->GetBitmapHandle());
#endif
*OpenCV 관련 라이브러리에 대한 에러 처리 방법 (데모 기준)
https://petra.tistory.com/929
https://whyprogrammer.tistory.com/525
Conference Call 진행시 화상 화면 배치 검토 건입니다.
현재는 2행 다열구조로 되어 있습니다.
검토시 추가 반영해야 할 사항은 사람이 제거됨에 따라서 영상 배치가 어떻게 되지는 확인이 필요합니다.
현재 서버에서 공유한 algorithm(codec & encryption)에 따른 동적 적용에 대한 코드 미 적용 상태로 해당 사항에 대한 검토가 필요합니다.
[Local connection]
ssh [email protected]
PW : lg
[AP connection]
ssh [email protected]
PW : lg
Implementing a jitter buffer in the receiver
jitterbuffer 적용에 따른 network issue 대응( 동적 video quality변경 포함)
Check communication protocol of client/server via wireshark
Review implementation of GStreamer functions using specific filters of GStreamer
일반적으로 오디오 데이터의 출력 모듈은 안정적인 오디오 출력을 위해서 자체적인 버퍼를 운용하고 있다. 이로인해 rtpjitterbuffer에 예를들어 100ms의 버퍼사이즈를 설정 했더라도 오디오 출력모듈의 버퍼로 데이터가 흘러들어가기 때문에 이를 고려해서 rtpjitterbuffer의 사이즈를 설정해야 한다.
In general, an audio data output module operates its own buffer for stable audio output. Because of this, even if the buffer size of rtpjitterbuffer is set, for example, 100ms, data flows into the buffer of the audio output module, so the size of the rtpjitterbuffer should be set considering this.
Server / Client 가 다룰 Data structure 및, TCP 의 Message 명세를 정의하기.
<공유 Excel file>
https://docs.google.com/spreadsheets/d/1giDLdlJh2AHMPdwRwXJ3_lERgXubzX4h58TgA41D8OE/edit?usp=sharing
Ability to change resolution during video streaming
Encrypt client pw using md5
acoustic echo cancellation (AEC) 구현 검토 건
현재 demo code적용 검토중
아래 정의한 TCP 송수신 Message에 따라서 TCP 에 Data를 전송할수 있는지 Feasibility 검사하기
https://docs.google.com/spreadsheets/d/1giDLdlJh2AHMPdwRwXJ3_lERgXubzX4h58TgA41D8OE/edit?usp=sharing
기본 TCP 전송방식은 아래 코드를 기반으로 검토하자
https://github.com/codekistone/VoIP
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