Beyond TCP: Meet the Next Generation of Transport Protocols

From plain old transmission control protocol (TCP) to newly conceived protocols, 通过互联网传输视频的各种方法是整个流媒体行业感兴趣的关键领域. 毕竟, 如果交付方式跟不上，那么最好质量的捕获和压缩又有什么用呢?

Back in January, in an article called “延迟糟透了!” 这是为了降低交互式或流媒体视频的整体传输时间, 我们谈到了一些较新的传输协议衍生品:web实时通信(WebRTC)。, reliable user datagram protocol (UDP), and plain old real-time protocol (RTP). Lower latency typically equates with lower-quality compression, 假设给处理器压缩视频图像的时间越长, 质量越好. 在本文中, 我们将更深入地研究底层协议，并讨论哪些协议对特定的应用程序有意义. 而OTT (over-the-top)实时线性传输的基础依赖于可靠的协议, TCP, 实际上，流媒体专业人员可以使用许多技术.

With an eye on lowering latency and increasing reliability, 在很好地处理邻居(相邻的数据包和网络设备)的同时，现在让我们探索在TCP的基础上扩展或取代TCP作为流传输霸主的替代传输协议.

亦WebRTC

As a quick recap from January, here’s what we know about WebRTC. 由Google和其他开源和开放标准社区的主要参与者推动, WebRTC was designed for peer-to-peer communication, 主要是小团体.

WebRTC is designed to use UDP and RTP by default, al虽然 it can be set to use a TCP-based fallback if anomalies are detected. 不足为奇的是, the video codecs for WebRTC are VP8 and VP9, 谷歌在致力于开放媒体联盟AV1编解码器的同时，还在继续推进这一技术. AVC，更广为人知的是H.264, 也可以用, 尽管AVC的音频同伴不能:音频通常是一个叫做Opus的开源编解码器, 而不是更广泛使用的AAC.

WebRTC is able to achieve very low latency, but doesn’t normally operate well in a typical streaming environment, meaning one that is based on real-time messaging protocol (RTMP), HLS, 或者AVC和AAC.

Companies such as Nanocosmos GmbH, 这家总部位于柏林的公司专注于从媒体服务器到最终用户的解决方案, have put forth concepts to create a scalable WebRTC live scenario. But the company acknowledges that, 在交货方面, it is missing CDN and vendor support, including any support by Apple.

修复TCP

Many different approaches have attempted to “repair” TCP, 包括一些试图通过缩小窗口大小来规避其固有问题的方法, if a TCP packet has not been received by the end-user device, that device can request that the packet be retransmitted). If the TCP window is too short, 虽然, 当分组传输被备份或分组之间的冲突增加时，问题就随之而来了.

使用WebSockets是一种尝试创建持久传输状态(本质上是一个隧道)的方法，它可以消除由于TCP窗口时间过短而导致的TCP数据包传输错误的累积. 还有一些解决方案通过降低http传输视频(如HLS或MPEG-DASH)的段大小来实现，从而允许更快的启动时间和更低的总体延迟.

HLS和DASH都受到限制，需要基于文件的段，这些段是通过HTTP请求提取的,Oliver Lietz说, 的首席执行官 Nanocosmos. “Due to the nature of HTTP and internet connections, 在不牺牲性能和稳定性的情况下，不能轻易地将段大小减小到2秒以下.”

HLS的这些段传统上是MPEG-2传输流(M2TS或TS), 它本身是基于几十年前的异步传输模式(ATM)协议，该协议旨在通过卫星发送视频信号. Besides the time it takes to segment video into 22- to 10-second segments, 实际的TS封装具有相对较多的报头位以及交错音频.

这些报头位对于以直接1:1的链路从地面站到卫星再到接收卫星碟形天线的方式重组通过卫星发送的内容非常有用, 但在网络传输协议处理传输顺序的TCP环境中，它们是不必要的.

Advances made in packaging of segments were first addressed in late 2011, Adobe和微软联合提出了使用碎片化MP4文件的案例，这将允许传输多种视频流排列.g., camera angles) and audio streams (e.g., 替代语言或评论轨道)，而不需要基于对M2TS封装方法的依赖而减慢HLS的交错.

MPEG-DASH adopted the fragmented MP4 (fMP4) approach, as did Apple in the most recent version of its iOS mobile platform. 迁移到fMP4的一个结果是能够完全避免HLS和DASH基于文件段的限制.

例如, Nanocosmos uses frame-based segments from an MP4 live stream, essentially allowing fMP4 to act as the “segment” to achieve ultra-low latency, with a fallback to HLS low latency for standard HLS players.

恢复UDP

TCP的兄弟被称为UDP，它的设计不一定是为了与其他协议很好地配合.

很简单, 低级因特网协议, at least when compared to TCP, the UDP approach forgoes a specific handshake between sender and receiver. This helps with speed of delivery, 但是，由于数据包没有得到接收方的确认，因此无法保证交付.

“The market was thirsty for an open source, 免费提供, 低延迟, UDP-like approach for streaming over the internet,彼得·马格说, 首席营销官 Haivision, 该公司与Wowza共同开发了一种名为安全可靠传输(SRT)的协议，以融合UDP和TCP的优势.

“SRT blends the resilience of TCP/IP transmission with the performance of UDP,马格说。, ，并增加了安全感, network-health监控, and simplified firewall traversal.”

The market approach for SRT is not from media server to end user, but from ingest point to media server. 以这种方式, SRT is positioning itself to be a limited-scalability replacement for RTMP, a goal of WebRTC proponents as well.

“SRT is currently ideal for contribution and distribution of performance streams,马格说。, 并补充说, along with other open-source efforts, there is a goal “to extend SRT to address broad-scale OTT delivery challenges.”

Lietz thinks that RTMP is still the best approach for ingest, given the wide number of RTMP-enabled encoders on the marketplace. 他还认为UDP“潜在地降低了复杂性，并允许创建低延迟的应用程序.” But he adds a caveat: “For reliable live-streaming applications, several application layers need to be added on top of UDP.”

One approach is to add a transport stream on top of UDP, 就像几十年前M2TS格式将MPEG-2多路复用成段一样.

“另外, forward error correction (FEC) needs to be added,Lietz说, 承认这些添加使UDP在延迟方面接近TCP的阈值.

更令人担忧的是, while UDP is sometimes used in multicast applications, and can use standard codecs like H.264和AAC, 就浏览器支持而言，UDP没有普遍可用的应用程序标准.

Haivision’s Maag says that SRT addresses those issues. 除了, since SRT has been around for quite some time, he says it has found traction for ingest among certain niches.

“SRT可以被任何需要低延迟视频流硬件或软件的开发人员使用,马格说。, 他指出，SRT于2013年推出，“目前已被数百家顶级广播公司和企业用于性能流媒体应用.”

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