webrtc

EasyRTC在实时性要求极高的应急类领域的应用

由 耗尽温柔 提交于 2020-12-27 10:01:05
一、需求背景 5G网速带来的提升，也带来了两个很大的优势，一个是传输速度更快了，另一个就是网络容量更加大了；这将有效提升实时音视频等多媒体应用的发展速度。在5G视频的未来应用领域里面，例如：无人驾驶、远程看护、远程辅导、金融双录、保险理赔等等，对于网络的要求都很高，但仅仅是网络变的更好，变得像内网，并不代表音视频产品的体验就变的更好。 5G已经开始商用，网络带宽优势变得更大，也不再成为系统性能的瓶颈，公网变成了内网，但这并不能改变传输效果，试想一下，我们在内网测试系统的时候，效果真的有那么流畅吗？ 二、RTC的实际应用 我们有时候在研发测试系统的时候，协议的选择也是非常关键的一环，每一种协议的侧重点不一样，比如RTP侧重RealTime，HTTP侧重Web友好，HLS侧重SmoothStreaming，所以，不同的行业应用要选择对应的视频传输协议； WebRTC已经发展了很多年，从早期的网页视频会议，现在几乎可以理解：WebRTC=Web + RTC，RTC也就是即时通信，WebRTC超越Web，可以应用于非常多的应急指挥的场景： 消防领域 消防无人机实时视频具有机动快速、使用成本低、维护操作简单等技术特点，具有对地快速实时巡察监测能力，是一种新型的中低空实时视频成像和红外成像快速获取系统。消防无人机通过摄像头将远程的火灾视频发回到指挥中心，让消防战士了解真实的火灾状况

WebRTC - How to identify the source IDs (device IDs) for a MediaStream(Tracks)

由 怎甘沉沦 提交于 2020-12-25 16:30:47
问题 Once I have established a WebRTC connection, I can getLocalStreams and getRemoteStreams . The streams have their own ID. Each stream is made of audio/video tracks MediaStreamTrack . I'd like to be able to identify the device/source (ID) used for each track in order to avoid adding a stream that was already added to the connection. In other word, I was thinking to make a check in connection.onaddstream that the stream track is not already present in my list of local or remote streams. Any

WebRTC - How to identify the source IDs (device IDs) for a MediaStream(Tracks)

由 南笙酒味 提交于 2020-12-25 16:15:10
问题 Once I have established a WebRTC connection, I can getLocalStreams and getRemoteStreams . The streams have their own ID. Each stream is made of audio/video tracks MediaStreamTrack . I'd like to be able to identify the device/source (ID) used for each track in order to avoid adding a stream that was already added to the connection. In other word, I was thinking to make a check in connection.onaddstream that the stream track is not already present in my list of local or remote streams. Any

WebRTC - How to identify the source IDs (device IDs) for a MediaStream(Tracks)

由 狂风中的少年 提交于 2020-12-25 16:14:54
问题 Once I have established a WebRTC connection, I can getLocalStreams and getRemoteStreams . The streams have their own ID. Each stream is made of audio/video tracks MediaStreamTrack . I'd like to be able to identify the device/source (ID) used for each track in order to avoid adding a stream that was already added to the connection. In other word, I was thinking to make a check in connection.onaddstream that the stream track is not already present in my list of local or remote streams. Any

WebRTC - How to identify the source IDs (device IDs) for a MediaStream(Tracks)

由 爱⌒轻易说出口 提交于 2020-12-25 16:14:23
问题 Once I have established a WebRTC connection, I can getLocalStreams and getRemoteStreams . The streams have their own ID. Each stream is made of audio/video tracks MediaStreamTrack . I'd like to be able to identify the device/source (ID) used for each track in order to avoid adding a stream that was already added to the connection. In other word, I was thinking to make a check in connection.onaddstream that the stream track is not already present in my list of local or remote streams. Any

WebRTC - How to identify the source IDs (device IDs) for a MediaStream(Tracks)

由 偶尔善良 提交于 2020-12-25 16:13:08
问题 Once I have established a WebRTC connection, I can getLocalStreams and getRemoteStreams . The streams have their own ID. Each stream is made of audio/video tracks MediaStreamTrack . I'd like to be able to identify the device/source (ID) used for each track in order to avoid adding a stream that was already added to the connection. In other word, I was thinking to make a check in connection.onaddstream that the stream track is not already present in my list of local or remote streams. Any

How to access Camera and Microphone in Chrome without HTTPS?

由 泄露秘密 提交于 2020-12-24 23:51:33
问题 Is there any way to access the Microphone and Camera using chrome when the website is http? I tried enabling "Insecure origins treated as secure" flag, however when the browser is closed and opened again, the domain used in this flag disappears and I have to retype and re-enable it again. I also tried starting chrome with "--unsafely-treat-insecure-origin-as-secure=http://example.com" argument, however when this was used a message appears on Chrome saying You are using an unsupported command

How to access Camera and Microphone in Chrome without HTTPS?

由 被刻印的时光 ゝ 提交于 2020-12-24 23:49:04
问题 Is there any way to access the Microphone and Camera using chrome when the website is http? I tried enabling "Insecure origins treated as secure" flag, however when the browser is closed and opened again, the domain used in this flag disappears and I have to retype and re-enable it again. I also tried starting chrome with "--unsafely-treat-insecure-origin-as-secure=http://example.com" argument, however when this was used a message appears on Chrome saying You are using an unsupported command

UseEffect hook with socket.io state is not persistent in socket handlers

由 吃可爱长大的小学妹 提交于 2020-12-23 00:50:11
问题 I have the following react component function ConferencingRoom() { const [participants, setParticipants] = useState({}) console.log('Participants -> ', participants) useEffect(() => { // messages handlers socket.on('message', message => { console.log('Message received: ' + message.event) switch (message.event) { case 'newParticipantArrived': receiveVideo(message.userid, message.username) break case 'existingParticipants': onExistingParticipants( message.userid, message.existingUsers ) break

【技术方案】一对一或一对多音视频通话会议系统，可以通过哪些方式实现？

由 夙愿已清 提交于 2020-12-22 07:55:19
讲到音视频互动通话系统，目前市面上出现比较多的有两种，一种是基于WebRTC的视频互动系统，另一种是RTMP/RTSP的一对一互动。由于近期TSINGSEE青犀视频研发团队在对这个做研究，因此我们也和大家探讨对比一下这两个互动系统。 WebRTC 很多互动系统都是基于webrtc做的，其最大的优点就是方便。现在WebRTC技术内置于浏览器中，用户不需要使用任何插件或者软件就能通过浏览器来实现实时通信。对于开发者来说，在Google将WebRTC开源之前，浏览器之间实现通信的技术是掌握在大企业手中，这项技术的开发是一个很困难的任务，现在开发者使用简单的HTML标签和JavaScript API就能够实现Web音/视频通信的功能。 但是在视频流传输方面，webrtc也有其不可避免的缺陷，其服务器部署非常复杂，不利于私有部署，在一些私密性高的场景下，无法使用，如公安、市政等体系；传输基于UDP，很难保证传输质量，由于UDP是不可靠的传输协议，在复杂的公网网络环境下，各种突发流量、偶尔的传输错误、网络抖动、超时等都会引起丢包异常，都会在一定程度上影响音视频通信的质量；且整个框架体系也不够灵活。 RTMP/RTSP 基于现有RTMP、RTSP推送、或内置RTSP服务、播放成熟的开发包，产品稳定度高，行业内首屈一指的超低延迟特性； 加入噪音抑制、回音消除、自动增益控制等特性，确保通话效果
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