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| author | Peter Bengtsson <mail@peterbe.com> | 2020-12-08 14:40:17 -0500 |
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| committer | Peter Bengtsson <mail@peterbe.com> | 2020-12-08 14:40:17 -0500 |
| commit | 33058f2b292b3a581333bdfb21b8f671898c5060 (patch) | |
| tree | 51c3e392513ec574331b2d3f85c394445ea803c6 /files/zh-cn/glossary/webrtc | |
| parent | 8b66d724f7caf0157093fb09cfec8fbd0c6ad50a (diff) | |
| download | translated-content-33058f2b292b3a581333bdfb21b8f671898c5060.tar.gz translated-content-33058f2b292b3a581333bdfb21b8f671898c5060.tar.bz2 translated-content-33058f2b292b3a581333bdfb21b8f671898c5060.zip | |
initial commit
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| -rw-r--r-- | files/zh-cn/glossary/webrtc/index.html | 29 |
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diff --git a/files/zh-cn/glossary/webrtc/index.html b/files/zh-cn/glossary/webrtc/index.html new file mode 100644 index 0000000000..73bcfa8530 --- /dev/null +++ b/files/zh-cn/glossary/webrtc/index.html @@ -0,0 +1,29 @@ +--- +title: WebRTC +slug: Glossary/WebRTC +tags: + - JavaScript + - Web + - WebRTC +translation_of: Glossary/WebRTC +--- +<p><span class="seoSummary"><strong>WebRTC</strong> (<em>Web Real-Time Communication</em>) 是一个可以用在视频聊天，音频聊天或P2P文件分享等Web App中的 {{Glossary("API")}}。</span></p> + +<p>WebRTC主要由以下几个部分组成：</p> + +<dl> + <dt><a href="/en-US/docs/Web/API/Navigator/getUserMedia"><code>getUserMedia</code></a></dt> + <dd>为一个RTC连接获取设备的摄像头与(或)麦克风权限，并为此RTC连接接入设备的摄像头与(或)麦克风的信号。</dd> + <dt><a href="/en-US/docs/Web/API/RTCPeerConnection"><code>RTCPeerConnection</code></a></dt> + <dd>用于配置音频或视频聊天。</dd> + <dt><a href="/en-US/docs/Web/API/RTCDataChannel"><code>RTCDataChannel</code></a></dt> + <dd><font face="consolas, Liberation Mono, courier, monospace">用于设置两个浏览器之间的</font>{{Glossary("P2P", "端到端")}} 数据连接。</dd> +</dl> + +<h2 id="了解更多">了解更多</h2> + +<ul> + <li>{{Interwiki("wikipedia", "WebRTC")}} on Wikipedia</li> + <li><a href="/en-US/docs/Web/Guide/API/WebRTC">Guide to WebRTC on MDN</a></li> + <li><a href="http://caniuse.com/rtcpeerconnection">Browser support of WebRTC</a></li> +</ul> |