WebRTC 1.0 is almost done. Now that it has reached the Candidate Recommendation stage, work has begun in earnest on the next version of WebRTC. In this talk, Dan covers the history of the WebRTC Standard and then presents the ideas that are being discussed for the next version of the standard.

WebRTC 1.0 concludes the mission to offer developers access to high-quality audio, video and data communication using a common set of APIs and protocols in browsers, mobile platforms, and IoT devices. The speech provides an overview of the many components, standards and technical challenges covered by WebRTC and how it has transformed the communications industry. You will learn more about what Google is doing to make it easy to develop rock-solid RTC services that work in any browser with new browser technologies, testing tools and standard support. Lastly the talk will give a sneak peak of what is coming is to WebRTC after 1.0.

演讲从互联网互动直播的商业应用，营收及成本模式入手，讨论互动直播相对于点播（VOD）以及传统线性电视（地面电视、有线电视、IPTV）在播放平台、后台架构、技术要求等诸多纬度的差异。基于这些特殊性连带上游视频编码行业近期的剧烈下滑，互动直播平台对于上线新编码格式面临着前所未有的技术挑战。

演讲介绍了 Cocos 引擎当前的技术进展、最战略布局和市场定位，覆盖了手机游戏出海、小游戏和区块链游戏等方面的内容，并阐释这些 Cocos 技术方案如何帮助开发者在2018年游戏行业面临诸多困境的情况下突围。

移动游戏存在外挂、被破解、重打包、恶意代码植入等安全问题，这些安全问题为企业带来了极大的隐患，玩家流失、玩家付费意愿降低、游戏失去公平性、研发成本提高等，最终导致企业收入减少、投入增加。

2018年WebRTC迎来了暴发，TutorABC团队成了第一批国内试水者，在此过程中我们走了许多弯路，经历了许多痛苦，也收获了许多成果，接下来我将通过以下几个维度给大家分享其中的一些经验。1、WebRTC在大规模直播场景下的架构演进2、WebRTC在前端页面中的交互实现3、WebRTC在前端应用中需要注意的问题，以及需要迈过去的坑

HTTP协议虽没有限制只允许使用TCP传输，但一直以来都是依赖于TCP实现，这是事实标准。随着基于UDP之上构建的QUIC传输层协议的出现，其天生内嵌HTTP 2协议的支持，让我们在对外开放HTTP API时多了一种传输通道的选择。微博移动产品部团队在将QUIC应用在移动端生产实践环境的实践过程中，通过在客户端和服务器端做出一些定制，比如QUIC直连、线上环境支持1-RTT和0-RTT握手优化机制等，逐渐摸索出适应自身的实际业务需求。在具体实践中，我们在QUIC规模部署中也踩了一些坑，定制开发了QUIC性能压测工具用于线上容量评估等，针对不同的实际场景我们的降级策略等等行为，在本次议题中，我们将完整的实践内容和反思，一一分享给大家。

Real-time communications have been evolving and going mainstream with the improvement and appearance of new open source technologies that make the development more affordable. Due to the continuous internet speed and processing power growth, the usage of applications to deliver real time multimedia both using mobile and desktop has become easier to implement. Today, being everywhere and connecting with everyone or everything is possible, and WebRTC is responsible for it.

Is WebRTC the only way to go if you want real-time communications in Web? As it usually happens, the answer is no, there are other options. When does using WebRTC makes sense? Where is WebRTC now? What are its use cases and in which fields is it more commonly used? Definitely we still see the common multiparty video conferencing applications but there are many others. New technologies are here: AR, VR, IoT and AI. And all of them have use cases where real time is required. Is WebRTC going to stay with us in the future? What are those new use cases? What are its challenges?

Those are just some of the questions that we will analyze based on real survey data and experiences building WebRTC applications for many different customers with companies all around the globe.

每个实时通信产品都期望可以提供最高质量的媒体流传输。那对于终端用户来讲是最理想的状态，这些用户都处于不同的地理位置、终端设备性能和网络环境。本次演讲将和大家一同讨论在利用 AI 改进媒体流传输质量方面，Callstats.io 的经验。

玩家的网络体验是游戏运营中的重要部分，尤其是对联网对战类的游戏，因此网络质量的管控不可或缺。本次我们通过时间、地点、人物、事件四个维度，分析游戏中真正能够反映玩家网络体验的网络质量指标数据，实现对这些网络指标数据的收集、整理、分析，提升网络问题处理的响应时间，预防性的解决网络上的故障隐患，同时实现对公有云网络质量的管控；通过数据的积累，结合业界优秀的解决方案，对网络能力进行提升，改善玩家的游戏体验。