In 搜索 of Better Encoding Quality for WebRTC
Suffice it to say that WebRTC is finally out of kindergarten and moving into the elementary grades. 到底是哪个等级? 好吧, that likely depends on which web browser you’re using and which server technology or platform 你的 WebRTC implementation uses.
大多数 我的客户项目到目前为止使用 WebRTC ingest (or publishing) via a web browser, in which JavaScript application programming interfaces can capture audio and 从视频eo from locally connected devices like a webcam and microphone. 像这样, the 从视频eo 质量 of the live stream is relying on the browser vendor’s 编码 implementation, 这可能 利用你的GPU或CPU. (提示:在Chrome上，你 可以使用URL chrome: / / gpu 为了获得统计数据 你的 local machine’s GPU capabilities as supChrome移植.) Browser-based 编码 for WebRTC has limitations, specifically with variable frame sizes, frame rates, and bitrates while publishing. 这些可变性会造成严重后果 havoc on server-side recordings or within 从视频eo-switching software that is able to consume WebRTC提要.
Of these limitations, the 从视频eo bitrate is most troubling when it comes to high-质量 从视频eo. I find that WebRTC streams published by current web browsers have an upper limit for 从视频0比特率，通常在2Mbps左右. In Chrome, you can monitor the outbound bandwidth by tapping the real-time reports generated by the URL chrome: / / webrtc-internals. 试一试 你的self anytime you’re using a local capture of 你的 webcam in a browser-based WebRTC publish荷兰国际集团(ing)会话. 你可能会看到类似的结果gardless的 which frame rate and resolution you choose for the capture. 如果你按同样的键eo content with a non-browser encoder solution, such as Millicast’s OBS WebRTC software (a free download, based on the same code as the original OBS software), 你可以取得更高的成就 质量. This OBS variant can push to Millicast’s platform as well as any server running Janus Web-RTC服务器，它是开源的.
举个例子， 图1 (下图) 实时显示 bitrate graph of a Chrome-captured WebRTC publish荷兰国际集团(ing)会话 for a virtual webcam and mic driven by Telestream’s Wirecast into Millicast的平台. Here, the 从视频eo bitrate is averaging 2Mbps.
图1. Chrome编码会话
图2(下面) 播放同样的视频 content being streamed directly via WebRTC into Millicast with its OBS WebRTC software, 这是使用FFmpeg和x264 Hood对内容进行实时编码. (Note that OBS can also utilize the GPU for NVENC 编码.)设置OBS的输出设置 to 8Mbps for the 从视频eo bitrate, and the results in 图2 show an average 8Mbps bitrate be用于Web-RTC播放.
图2. OBS WebRTC编码会话
It’s beyond the scope of this column to discuss the implications of simulcast WebRTC on 编码 performance within software solutions such as OBS, but bear in mind that both client- and server-side WebRTC solutions can assume the role of pro从视频ing tiered resolutions 用于自适应流播放.
What do we do in the meantime if there isn’t an option to use a WebRTC publisher for a WebRTC播放场景? 许多Web-RTC平台as-a-service (PaaS) vendors and streaming server products will accept Real-Time Messaging Protocol (RTMP) ingest sessions and transcode AAC audio to WebRTC-compatible 作品音频. 像这样, you can achieve higher-质量 WebRTC playout streams by 编码 与任何现有的RTMP编码解决方案.
在其中之一 去年的专栏(go2sm.com/web-rtcproblem)，我强调了a的缺失 consistent signaling approach among WebRTC server vendors and made a plea for WebRTC vendors to implement a standard connection 设置. I’ve since seen WHIP (WebRTC HTTP Ingestion Protocol), a proposal from CoSMo Software’s Alex Gouaillard and Sergio Garcia Murillo. WHIP offers the potential to unify the next 波 real-time 编码 solutions by pro从视频ing an alternative and improvement to RTMP, allowing a wider range of audio and 从视频eo codecs. 希望，流行的产品在现场 从视频eo-switching software space, such as Wirecast, vMix, Vimeo Studio 6, Boinx Software’s 的主分支 OBS, will incorporate WHIP to give every web broadcast有机会使用Web-RTC. On the ingest side, I expect to see similar adop更多的WebRTC PaaS供应商和我Dia服务器产品.
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