Low-Latency Video Transport Technology

With the latency concerns raised with the used of HTTP streaming solutions, this post will discuss a few of the technologies that are being used to address low latency.

Secure Reliable Transport (SRT)

SRT is an open-source protocol, developed by Haivision, which allows broadcasters and streamers to deliver high-quality, low-latency video streams across the public internet. SRT is fast becoming the de facto low latency video streaming standard in the broadcast and streaming industries (Haivision, 2018).

Haivision originally introduced the technology to the industry in 2013 enabling low latency video transport across the television production and broadcast market. In April 2017, Haivision made the SRT protocol available to the world as open source technology and along with Wowza formed the SRT Alliance (Haivision, 2018).

The short video below from Haivision explains what SRT means.

SRT is referred to with this formula; Low Latency + High Quality + Security = SRT. It is designed to address video streaming shortcomings like packet loss, jitter, delay and bandwidth issues (Wowza.com, 2018). Figure 1 below is an illustration of SRT Open Source Interoperability.

Figure 1 - SRT in action (Source: matrox.com)

During a special presentation at IBC 2018, a panel of experts from ViaCom, Al-jazera, SkyNews and others spoke about the benefits of SRT and how it helps streamline workflows (Regan, 2018), they include

• Pristine quality video: SRT is designed to protect against jitter, packet loss, and bandwidth fluctuations due to congestion over noisy networks for the best viewing experience possible. This is done through advanced low latency retransmission techniques that compensate for and manage the packet loss. SRT can withstand up to 10% packet loss with no visual degradation to the stream.
• Low latency: In spite of dealing with network challenges, video and audio is delivered with low latency in SRT systems. It has the combined advantages of the reliability of TCP/IP delivery and the speed of UDP. 
• Secure end-to-end transmission: Industry-standard AES 128/256-bit encryption ensures protection of content over the internet. SRT provides simplified firewall traversal. 
• Leveraging the internet: Because SRT ensures security and reliability, the public internet is now viable for an expanded range of streaming applications—like streaming to socialcast cloud sites (for example, LiveScale omnicast multi-cloud platform's concurrent distribution to multiple social media such as Facebook Live, YouTube, Twitch and Periscope from one live video feed), streaming or remoting an entire video wall content, or regions of interest of a video wall, and more. 
• Interoperability: Users can confidently deploy SRT through their entire video and audio streaming workflows knowing that multi-vendor products will work together seamlessly.
• Open source: Royalty-free, next-generation open source protocol leads to cost-effective, interoperable, and future-proofed solutions. 

The possibilities are endless for what can be achieved with SRT in the streaming technology industry.

Web Real-Time Communication (WebRTC)

The Web Real-Time Communication (WebRTC) framework is an open source project started by Google in 2011 that enables plugin free real-time communication of audio, video and data in Web and native apps.

Figure 2 - WebRTC framework (Source: wowza.com)

The framework was developed with a peer-to-peer architecture in mind, where a small group of clients can communicate directly with each other. WebRTC employs three HTML5 APIs that are built into the Chrome, Firefox and Safari browsers. These APIs turn the browsers into encoders for video streaming, which then connect directly to other browsers for playback (Siglin, 2018).

In using this protocol however, the user needs to determine which situations are best suited for it. The next sections details 2 scenarios where the use of WebRTC can be employed and a further 2 where it cannot be used due to restrictions.

When it can be used:

• Video Conferencing or Real Time Interactivity: WebRTC is a good option for group video conferencing and interactive cases with smaller audiences. To achieve this, WebRTC needs to be used for publishing and playback, as it supports real-time communication (less than 0.5 seconds) in good network conditions. This would ensure nearly concurrent interactions.
• You need to build and/or broadcast through a web app with minimal infrastructure: Since WebRTC uses HTML5 APIs, this allows access to the features available in the HTML5 programming language. These features allow users to make use of the functional offerings without plug-ins. With this, users can stream videos from their browsers and even share screens to audiences with no additional infrastructure and plug-ins (Balistreri, 2018).

When it cannot be used:

• Streaming at Scale or Viral viewership: Currently, WebRTC is limited in its ability to scale without a network of live-repeating servers to handle the load. Since WebRTC uses peering network this would not be possible.
• You need broadcast-quality streaming: Broadcast quality streaming has to be of the highest quality possible and at current, WebRTC does not support that. The bitrate, speed and level of connectivity needed to achieve broadcast quality streaming cannot be attained in WebRTC. The protocol is currently limited to supporting VP9 and .H264 video, anything more would bog down the network (Balistreri, 2018).

WebRTC is an amazing and highly disruptive standard that involves the orchestration of many technologies and protocols. Both desktop and mobile-based multi-person multimedia chat applications are fully achievable by leveraging WebRTC.

References

Balistreri, A. (2018). What Is WebRTC? | Low-Latency Streaming | Wowza. [online] Wowza Product Resources Center. Available at: https://www.wowza.com/blog/what-is-webrtc [Accessed 20 May 2019].
Haivision. (2018). SRT (Secure Reliable Transport) Protocol | Haivision. [online] Available at: https://www.haivision.com/products/srt-secure-reliable-transport/ [Accessed 22 May 2019].

Haivision. (2019). What is Latency for Video? - Definition of Video Latency | Haivision. [online] Available at: https://www.haivision.com/resources/streaming-video-definitions/video-latency/ [Accessed 19 May 2019].


InnoArchiTech. (2018). What Is WebRTC and How Does It Work?. [online] Available at: https://www.innoarchitech.com/what-is-webrtc-and-how-does-it-work/ [Accessed 20 May 2019].

Regan, H. (2018). Secure, Reliable, Low-Latency Video With SRT: IBC 2018 Panel | Wowza. [online] Wowza Product Resources Center. Available at: https://www.wowza.com/blog/secure-reliable-low-latency-video-with-srt-ibc-2018-panel-video [Accessed 20 May 2019].

Siglin, T. (2018). Latency Sucks Less: Advancements in Low-Latency Live Streaming | Wowza. [online] Wowza Product Resources Center. Available at: https://www.wowza.com/blog/latency-sucks-less-advancements-in-low-latency-live-streaming [Accessed 20 May 2019].

Wowza.com. (2018). SRT Secure Reliable Transport. [online] Available at: https://www.wowza.com/low-latency/SRT-secure-reliable-transport [Accessed 20 May 2019].