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WebRTC: Support WebRTC, for video meeting, SFU server, video converence. #307

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winlinvip opened this issue May 16, 2015 · 37 comments · Fixed by #2303
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WebRTC: Support WebRTC, for video meeting, SFU server, video converence. #307

winlinvip opened this issue May 16, 2015 · 37 comments · Fixed by #2303
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Feature It's a new feature. TransByAI Translated by AI/GPT.
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@winlinvip
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winlinvip commented May 16, 2015 •
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WebRTC is now quite mature, with stable playback and the protocol already being an RFC. There are also quite a few corresponding open-source projects. However, I believe that WebRTC still lacks a high-performance, simple and easy-to-use server. I have analyzed the existing servers before and found various issues. SRS has a great opportunity to solve these problems.

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@winlinvip winlinvip added the Feature It's a new feature. label May 16, 2015
@winlinvip winlinvip added this to the srs 3.0 release milestone May 16, 2015
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@winlinvip winlinvip changed the title 支持WebRTC Support WebRTC Dec 7, 2019
@winlinvip winlinvip mentioned this issue Dec 19, 2019
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winlinvip added a commit that referenced this issue Jan 19, 2020
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For #307, #1070, define FLV CodecID for AV1 and Opus. 3.0.101
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@winlinvip winlinvip mentioned this issue Jan 25, 2020
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winlinvip commented Mar 14, 2020 •
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candidate is the address that RTC provides services to the outside. In certain deployment scenarios, we need to configure it in multiple ways.

This is also the only item that RTC must confirm the configuration for, while others can be left at their default settings.

For detailed configuration instructions, please refer to: https://github.com/ossrs/srs/wiki/v4_CN_WebRTC#config-candidate

TRANS_BY_GPT3

winlinvip added a commit that referenced this issue Mar 14, 2020
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For #1638, #307, rtc conf support ENV.
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winlinvip added a commit that referenced this issue Mar 14, 2020
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For #1638, #307, refactor rtc config.
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winlinvip added a commit that referenced this issue Mar 14, 2020
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For #1638, #307, use webrtc adapter.js for demo.
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winlinvip added a commit that referenced this issue Mar 14, 2020
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For #1638, #307, use webrtc adapter.js for demo.
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winlinvip added a commit that referenced this issue Mar 14, 2020
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For #1638, #307, define webrtc:// url for play
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winlinvip added a commit that referenced this issue Mar 14, 2020
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For #1638, #307, merge john, add NACK suport, remove debug code, verb…
60c8b37 
…ose log
winlinvip added a commit that referenced this issue Mar 14, 2020
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For #1638, #307, fix rtc player demo bug
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winlinvip added a commit that referenced this issue Mar 15, 2020
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For #1638, #307, remove unused bash variable.
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winlinvip added a commit that referenced this issue Mar 16, 2020
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For #1638, #307, add comments in RTMP/RTP message for RTC.
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winlinvip added a commit that referenced this issue Mar 16, 2020
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For #1638, #307, add comments in RTMP/RTP message for RTC.
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winlinvip added a commit that referenced this issue Mar 16, 2020
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For #1638, #307, accept srs_string_dumps_hex to app utility.
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winlinvip added a commit that referenced this issue Mar 16, 2020
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For #1638, #307, add comments for api and enum
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winlinvip added a commit that referenced this issue Mar 17, 2020
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For #1638, #307, fix build warnings
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@21877575 21877575 mentioned this issue Jul 16, 2020
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@tyranron tyranron mentioned this issue Aug 15, 2020
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@jahnsoncade jahnsoncade mentioned this issue Feb 6, 2021
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@winlinvip winlinvip mentioned this issue Mar 24, 2021
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@tingfeng-key tingfeng-key mentioned this issue Mar 31, 2021
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@winlinvip winlinvip mentioned this issue Apr 19, 2021
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@winlinvip winlinvip linked a pull request Apr 27, 2021 that will close this issue
Support transmux RTC to RTMP. 4.0.95 #2303
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@ossrs ossrs locked as too heated and limited conversation to collaborators Apr 28, 2021
@winlinvip winlinvip pinned this issue May 16, 2021
@winlinvip winlinvip changed the title Support WebRTC Support WebRTC，一对一通话，多人通话，SFU，直播连麦，会议录制等 May 16, 2021
@winlinvip winlinvip changed the title Support WebRTC，一对一通话，多人通话，SFU，直播连麦，会议录制等 WebRTC: Support WebRTC, for video meeting, SFU server, video converence. 一对一通话，多人通话，SFU，直播连麦，会议录制等 Nov 25, 2022
@winlinvip winlinvip changed the title WebRTC: Support WebRTC, for video meeting, SFU server, video converence. 一对一通话，多人通话，SFU，直播连麦，会议录制等 WebRTC: Support WebRTC, for video meeting, SFU server, video converence. Jul 25, 2023
@winlinvip winlinvip added the TransByAI Translated by AI/GPT. label Jul 25, 2023
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Support transmux RTC to RTMP. 4.0.95 ossrs/srs