webrtc

Where to store WebRTC streams when building React app with redux

我与影子孤独终老i 提交于 2021-02-06 02:01:50
问题 I'm building a React.js application that interacts with the WebRTC apis to do audio/video calling. When a call is successfully established, an 'onaddstream' event is fired on the RTCPeerConnection instance, which contains the stream that I as a developer am supposed to connect to a video element to display the remote video to the user. Problem I'm having is understanding the best way to get the stream from the event to the React component for rendering. I have it successfully working by just

Where to store WebRTC streams when building React app with redux

匆匆过客 提交于 2021-02-06 02:00:05
问题 I'm building a React.js application that interacts with the WebRTC apis to do audio/video calling. When a call is successfully established, an 'onaddstream' event is fired on the RTCPeerConnection instance, which contains the stream that I as a developer am supposed to connect to a video element to display the remote video to the user. Problem I'm having is understanding the best way to get the stream from the event to the React component for rendering. I have it successfully working by just

How do peers involved in a p2p communication authenticate each other?

自闭症网瘾萝莉.ら 提交于 2021-02-05 12:22:04
问题 How do peers in WebRTC authenticate each other? 回答1: DTLS in WebRTC uses self-signed certificates. RFC 5763 has the details, in a nutshell the certificate fingerprint is matched against the one provided in the a=fingerprint line of the SDP. 回答2: As the comment from Patrick Mevzek already mentioned: It doesn't depend on DTLS or TLS, if a self-signed certificate is trusted/accepted. This depends only from the peer's trusted certificates. If the client's or server's certificate path/chain

WebRTC iOS: Record Remote Audio stream using WebRTC

我怕爱的太早我们不能终老 提交于 2021-02-04 19:27:49
问题 I am working on an audio streaming application with recording functionality for a receiver. I got stuck at the point where the user want to record audio stream on the receiver side. Below is my code Initialisation var engine = AVAudioEngine() var recordingFile: AVAudioFile? var audioPlayer: AVAudioPlayer? let player = AVAudioPlayerNode() var isRecording: Bool = false Initialise AudioEngine func initializeAudioEngine() { let input = self.engine.inputNode let format = input.inputFormat(forBus:

你女朋友在买买买时,程序员小哥在干嘛?

风流意气都作罢 提交于 2021-02-04 00:39:34
年货节来了,你女朋友下单了吗? 观看直播已经成为人们日常上网的习惯之一,直播购物作为其中的重要项目,销售额屡创新高。最近,天猫年货节刚刚开幕,淘宝直播中就涌入了不少的用户前来抢购,毕竟足不出户就可以买到全国甚至全球的年货,对热衷买买买的消费者们来说实在太方便了。 那么,直播抢购背后到底蕴藏着哪些技术?平台在开发网络直播系统时为什么如此看重CDN?在高并发直播的当下,CDN的技术人员都需要关注什么?阿里云Edge Plus的第3期云话题,边缘酱将为你讲讲关于直播抢购与CDN的那些事。 你关心的,就是云话题 接下来进入正题 云话题 | 第3期 关于直播抢购与CDN的那些事 特邀专家:卢日 阿里云高级技术专家,GRTN网络总设计和布道师,目前负责阿里云视频直播产品和流媒体实时加速平台研发。 一、互联网直播最关键的技术指标是什么? 想必大家都知道,“直播”已经是非常普遍的一种娱乐形式了,直播的及时性和互动性成为信息触达、互动沟通的新媒介。随着5G、超高清、VR等技术的发展,主播与观众的互动需要更加实时,“延时”这个指标愈发的重要。 高延时影响了直播互动体验,阻碍了直播在一些场景的落地,特别在电商直播,直播间的评论提问是观众和主播互动的一个重要手段,主播的实时互动反馈对直播间的活跃度和交易达成至关重要。 二、直播中的那几秒延时都分布在哪里? 我们剖析直播延时的分布

OpenAI将k8s扩展至7500个节点以支持机器学习;Graph Diffusion Network提升交通流量预测精度

爷,独闯天下 提交于 2021-02-02 10:42:44
开发者社区技术周刊 又和大家见面了,快来看看这周有哪些值得我们开发者关注的重要新闻吧。 Google研究院推出处理文本图像新框架TReCS OpenAI将k8s扩展至7500个节点以支持机器学习 Apache ECharts 5正式发布 WebRTC成为W3C与IETF正式标准 国内首个自主可控区块链技术体系“长安链”发布 京东开源PyTorch人脸识别工具包FaceX-Zoo AAAI 2021丨Graph Diffusion Network提升交通流量预测精度 AAAI 2021丨利用标签之间的混淆关系,提升文本分类效果 行 业 要 闻 1.Google研究院推出处理文本图像新框架TReCS 为创建一种能够在任何语言之间进行翻译的通用神经机器翻译系统,Google 研究人员研发了一种新框架,即 TReCS(Tag-Retrieve-Compose Synthesize system) 。通过改进图像元素的唤起方式以及迹线如何通知其位置,从而显着增强图像生成过程。该系统接受了超过250亿个示例的培训,具有处理103种语言的潜力。其功能使鼠标轨迹与文本描述对齐,并为提供的短语创建可视标签。该框架利用可控的鼠标轨迹作为细粒度的视觉基础,根据用户的叙述生成高质量图像。标记器被用来预测短语中每个单词的对象标签。 2.OpenAI将k8s扩展至7500个节点以支持机器学习 为了满足GPT

Unable to Screencast to Apprtc using Replay Kit Over Webrtc

感情迁移 提交于 2021-02-02 09:57:24
问题 Hello There, I am trying to screen broadcast with latest webrtc libraries but I keep getting following error: iOS 13.0 & above: Live broadcast has stopped due to Attempted to start invalid broadcast session. less than iOS13 but greater than iOS 12.0: Live broadcast has stopped due to:(null) I would really appreciate if anyone can answer my question. Thanks var peerConnectionFactory: RTCPeerConnectionFactory? var localVideoSource: RTCVideoSource? var videoCapturer: RTCVideoCapturer? func

Unable to Screencast to Apprtc using Replay Kit Over Webrtc

生来就可爱ヽ(ⅴ<●) 提交于 2021-02-02 09:57:01
问题 Hello There, I am trying to screen broadcast with latest webrtc libraries but I keep getting following error: iOS 13.0 & above: Live broadcast has stopped due to Attempted to start invalid broadcast session. less than iOS13 but greater than iOS 12.0: Live broadcast has stopped due to:(null) I would really appreciate if anyone can answer my question. Thanks var peerConnectionFactory: RTCPeerConnectionFactory? var localVideoSource: RTCVideoSource? var videoCapturer: RTCVideoCapturer? func

WebRTC remote video is shown as black

偶尔善良 提交于 2021-02-01 09:07:48
问题 While developing a WebRTC video chat application I have encountered receiving remote the video stream. The video stream blob is received, but the video is just black. I have gone through these answers and tried almost everything I could to get it to work https://stackoverflow.com/a/17424224/923109 Remote VideoStream not working with WebRTC ...... Globalvars.socket.on('call', function (signal) { if(!Globalvars.pc){ Methods.startCall(false, signal); } if(signal.sdp){ temp = new

WebRTC 系列之视频辅流

末鹿安然 提交于 2021-01-30 03:58:35
作者:网易云信资深客户端开发工程师 陶金亮 近几年,实时音视频领域越来越热,业界很多音视频引擎都是基于 WebRTC 进行实现的。本文主要介绍 WebRTC 在视频辅流上的需求背景以及相关技术实现。 WebRTC 中的 SDP 支持两种方案: PlanB 方案 和 Unified Plan 方案。早期我们使用多PeerConnection的 Plan B 方案中只支持一条视频流发送,这条视频流,我们称之为”主流”。目前我们使用单 PeerConnection 的 Unified Plan 方案,新增一条视频辅流,何为视频”辅流”?视频辅流是指第二条视频流,一般用于屏幕共享。 需求背景 随着业务的发展,一路视频流满足不了更多实际业务场景的需求,例如在多人视频聊天、网易会议以及其他在线教育场景下,需要同时发送两路视频流:一路是摄像头流,另一路是屏幕共享流。 但是,目前使用 SDK 分享屏幕时,采用的是从摄像头采集通道进行屏幕分享。在该方案下,分享者只有一路上行视频流,该场景中要么上行摄像头画面,要么上行屏幕画面,两者是互斥的。 除非实例一个新的 SDK 专门采集并发送屏幕画面,但实例两个 SDK 的方案在业务层处理起来十分麻烦且会存在许多问题,例如如何处理两个流间的关系等。 在 WebRTC 场景中,还存在一种可以单独为屏幕分享开启一路上行视频流的方案,并称之为“辅流