Guest post Events such as the Super Bowl, the World Cup or the Eurovision song contest thrive on the tension. That is precisely why they are lucrative for online media providers. But an average delay of the online stream compared to conventional television by around 30 seconds can destroy this experience: When a football fan sees his team’s player running for a penalty on their cell phone, it is extremely annoying to see TV on television-Comments to learn that the ball went into the goal. It is also over with the tension when the spectators already know the result of a dribble from the reactions in the public area via the Internet TV on the restaurant terrace.
A slower signal transmission compared to traditional television lowers the acceptance of online live TV. According to the State of Online Video 2019 study conducted by Limelight, every second viewer in Germany would watch more live sports online if the broadcasting were not slower than conventional TV. Even less can content with interaction – such as online gaming events, auctions, quizzes or bets – allow the next video sequence to be displayed too slowly. They also require the signals to be sent back quickly.
It can take a while for a live video to be processed and transmitted to the end device via the network. This is mainly due to the fact that the internet protocol HTTP TCP / IP was not originally intended for the broadcast of video content. The time it takes to encrypt a camera video and make it available on an online player or OTT device is known as live streaming latency. There are numerous ways to reduce residues, but no one-size-fits-all solution. The most suitable method results from the content and its specific requirements.
Shorter chunks can help
The chunk size can be adjusted to solve the problem: Before each video playback via the HTTP-TCP / IP protocol, a player stores three video segments – chunks – created by the encoder in the buffer. This will need time. With a chunk length of 10 seconds, 30 seconds have to be buffered, with six seconds corresponding to 18 seconds. A shorter video segment therefore speeds up the transmission: With a length of up to one second, end-to-end latency of only 6 seconds can be achieved. Common HTTP chunked protocols such as HLS and MPEG-DASH are supported. The method is suitable for content that does not require real-time playback of the signals or interactivity. But it also has its limits: Chunks that are too short can require repeated player buffering if the data stream is interrupted and the buffer is empty. Therefore, every workflow should be tested accordingly.
Lower latency thanks to CMAF encryption
The Common Media Application Format (CMAF) also speeds up the display of signals on the player. A uniform framework is used for storing the data in HLS and MPEG-DASH. This has the advantage that content only has to be saved and packed once. Broadcasters no longer have to create two separate data sets of the same audio and video data in order to use different end devices such as tablets, computers, smartphones or other end devices.
The chunk’s CMAF encryption reduces latency. When transmitting segments without CMAF chunk, the video samples are only output by the encoder and sent over the network when a complete segment has been created. The decoder then begins decoding. In contrast, a segment divided into CMAF chunks uses encoded video chunks, so-called video fragments. The coding of the medium is necessary to correctly describe the sections and signals the availability of smaller sections. The individual fragments can be transmitted before the entire video segment has been processed. As a result, the decoder also starts playing before the entire segment is received.
WebRTC for real-time broadcasts
Web Real-Time Communications (WebRTC) is an open source project for communication via browsers and mobile applications. WebRTC is operated by, , Opera and other major players. Originally used for web conferences, the technology now enables video broadcasts with large numbers of viewers. To do this, it uses the UDP network protocol, which is more effective than TCP / IP. Chunk segmentation of the data streams by the encoder and intermediate storage are eliminated.
WebRTC also enables content to be played on standard web browsers – plug-ins or special video players are unnecessary. The bit rate can be adjusted so that the viewer sees content in optimal picture quality even under variable network conditions. In addition, the technology offers bidirectional transmission of data in real time by the viewer or online gamers. The 2-way data channel thus enables a latency of less than a second and thus the interaction required for gaming.
To keep viewers happy and loyal, media and broadcast companies need the services and technologies of content delivery network (CDN) providers like Limelight Networks. They support the industry in delivering a high-quality viewing experience to every user and also remain competitive in the lucrative event area. In order to be able to provide appropriate streams, obstacles to web traffic must be removed and interference-free transmission guaranteed. The media delivery infrastructure must remain flexible, as users increasingly follow live events online and become more and more mobile. The challenge is to create an experience in TV quality and in real time through these channels.