Contribution and distribution protocols in Broadcast environments

An overview of the protocols for TV signal contribution in broadcast environments: ST2110, SRT, NDI and more

By Yeray Alfageme,  Business Development Manager at Optiva Media an EPAM company

 

In the exciting world of broadcast, where innovation and technology are advancing by leaps and bounds, keeping up with the latest advances in contribution protocols and signal distribution is critical. In this article we will explore some of the most relevant protocols in broadcast environments, such as ST2110, SRT, NDI, DASH, HLS, RTMP and many more.

These protocols play a key role in the transmission and delivery of audiovisual content in real time, whether for traditional television, Internet transmission or video-on-demand services. Each of them has its own features and benefits, and it is important to understand how they work and when it is most appropriate to use each of them.

We’ll start by talking about the ST2110 standard, which has revolutionized the way in which video, audio and data signals are handled in IP television production and distribution environments. We will explore its architecture and how it enables transmission of multiple medium and high-quality streams in real time.

We will then dive into the SRT (Secure Reliable Transport) protocol, which has gained popularity due to its ability to ensure a safe and reliable delivery of video signals over the Internet. We will see how SRT uses error correction and selective broadcast techniques to ensure robust, low-latency transmission.

We will continue our journey with NDI (Network Device Interface), a protocol developed by NewTek that allows the transmission of high quality video and audio through a local area network (LAN). We will explore its applications in live production environments and how it has simplified connectivity and interoperability between different devices.

In addition, we will analyze the DASH (Dynamic Adaptive Streaming over HTTP) and HLS (HTTP Live Streaming) content distrivution protocols, widely used in video transmission over the Internet. We’ll look at how these adaptive streaming technologies work and how they’ve improved the viewing experience for end users.

We cannot fail to mention the popular RTMP (Real-Time Messaging Protocol), widely used for the real-time transmission of multimedia content over the Internet. We will explore its architecture and relevance in the current context of live video streaming.

 

ST2110

The ST2110 protocol has played a revolutionary role in changing the way video, audio and data signals are managed in IP-based broadcast contribution and distribution environments. ST2110 is a standard developed by SMPTE (Society of Motion Picture and Television Engineers) that defines a set of technical specifications for transport of real-time media streams over IP networks.

One of its main features is an approach based on separate streams, which means that video, audio and data are transmitted as separate streams, unlike traditional methods that used to encapsulate all components in a single stream. This provides greater flexibility and scalability, as each element can be handled and processed individually.

The ST2110 standard uses the IP (Internet Protocol) transport protocol for the routing and transmission of streams. This allows the interconnection of equipment from different manufacturers and the integration of systems into a common network infrastructure. In addition, the protocol leverages standard Ethernet networks, reducing deployment costs and facilitating migration to IP environments.

In terms of signal quality, ST2110 offers excellent fidelity and uncompressed accuracy. This protocol supports video resolutions up to 4K and beyond, as well as high-quality audio, including tools for immersive formats such as surround sound. In addition, ST2110 achieves low latency, which is critical in real-time applications such as live broadcasts or interactive productions.

ST2110 is designed for production environments. Due to its orientation towards uncompressed environments where large bandwidths are required -for example more than 1 Gbps in HD-SDR signals- it is practically unfeasible for content distribution. Almost the entire production environment in IP works with this protocol and it is true that -especially in large events and productions- it can be used to contribute between production centers over large-bandwidth private networks, but it is not recommended or at least appropriate.

In addition, ST2110 integrates effectively with other protocols and standards that are common in the industry, such as SMPTE ST2022-6 for transmission over IP networks and AES67 for audio interoperability over IP. This encourages compatibility and collaboration between different systems and ensures flexibility in the choice of equipment and technologies, at all times within a controlled, high-capacity network environment, such as a mobile unit or production center.

 

SRT (Secure Reliable Transport)

SRT is an open-source transport protocol developed by Haivision and specifically designed to overcome the challenges of streaming video over the Internet. With its combination of error correction and selective retransmission techniques, it achieves secure and reliable delivery of video signals, even in environments with packet loss, high latencies or fluctuations in bandwidth.

One of its outstanding features is the ability to dynamically adapt to network conditions in real time. By using smart flow control and congestion control algorithms, this protocol can automatically adjust transmission rate and amount of error correction according to network conditions. This ensures optimal transmission quality and minimizes the effects of packet loss.

Another important advantage is its focus on safety. The protocol uses end-to-end encryption to protect the confidentiality and integrity of the transmitted data, which guarantees privacy and prevents possible attacks or tamperingwith the signal. In addition, SRT includes authentication and access control mechanisms, which make it possible to verify the identity of the participants in the transmission and control who can access it.

SRT is primarily used in content distribution or in contribution between distant production sites where connectivity is achieved via the Internet. In these environments, the SRT facilitates the delivery of broadcast content to multiple destinations such as TV stations, Internet streaming service providers or video-on-demand platforms. The protocol is easily integrated into existing distribution systems and adapts to different bandwidth and latency requirements, thus making it ideal for live streaming environments or sporting events.

Furthermore, SRT has become a widely adopted industry standard, supported by numerous equipment manufacturers and service providers. This guarantees interoperability and compatibility between different solutions and ensures flexibility in the choice of equipment and technologies.

 

NDI (Network Device Interface)

Developed by NewTek -now Vizrt- NDI is a protocol that enables the transmission of high-quality video and audio, as well as ancillary data, over an IP network in its broadest sense. NDI is designed -like ST2110- for production environments but, due to the compression it performs on the signal, it can be used to distribute signals in certain controlled environments although, as we have commented, its main aim is to simplify connectivity between different devices in a live production environment and facilitate interoperability between them.

One of the main advantages of NDI is its software-based approach, which means it can be deployed on existing systems without the need for additional hardware. Thanks to NDI, any compatible device or application can become a video/audio source or destination on the network, which provides great flexibility and ease of use.

In addition, it is intended for use in standard Ethernet networks, due to its adaptation to existing bandwidths and technical features. One feature that exemplifies this fact is that an HD-SDR signal takes up less than 1 Gbps of bandwidth, making it possible to use a single Ethernet connection between the piece of equipment and the network.

The NDI protocol also offers low latency, which is crucial for live production and real-time collaboration. This means that changes and adjustments made to a video or audio source are quickly reflected at destinations, with no noticeable delays. This feature is especially valuable in production environments where synchronization and real-time response are critical issues.

As we have pointed out, the NDI protocol is geared towards contribution, more specifically to production environments. Its use in distribution environments is not as common, although it is possible, since it does not have such advanced error correction mechanism or adaptation to variable bandwidths as SRT or others.

NDI’s compatibility with a wide range of devices and software has contributed to its growing popularity in the broadcast industry. Many equipment manufacturers and software developers have adopted the NDI protocol in their products, thus ensuring interoperability and compatibility between different solutions.

 

RIST (Reliable Internet Stream Transport)

Developed by the RIST Initiative -which brings together multiple industry players- RIST is a transport protocol that uses the Internet for contribution and distribution of multimedia signals in real time. Its main goal is to overcome the challenges associated with transmission over the Internet by guaranteeing reliable and low-latency delivery.

One of its key features is the focus on transmission reliability. This protocol uses error correction techniques such as selective retransmission and lost packet recovery to ensure the integrity of the transmitted data. This allows robust transmission even in networks with packet loss, variable latency or congestion, which is very common in the Internet.

In addition, the RIST standard implements flow control and congestion control mechanisms to ensure efficient and adaptive transmission. These mechanisms monitor and adjust the amount of data transmitted based on network conditions, thus avoiding saturation and bottlenecks. This ensures optimal use of available bandwidth and minimizes the effects of network congestion.

Security is another major concern in broadcast transmission and RIST also addresses this aspect. The protocol offers end-to-end encryption options to protect the confidentiality of the transmitted data. In addition, it includes authentication and access control mechanisms, which make it possible to verify the identity of the participants in the transmission protect it against malicious attacks.

As its name suggests, RIST is 100% oriented to the distribution of content through the Internet. Of course, it can be used in contribution and production environments, but it was not developed for this. This protocol is especially reliable, perhaps with a higher latency than similar distribution protocols, but with extra security and robustness.

It should be mentioned that RIST, although quite widespread -that’s why we include it here- is not as popular as more modern protocols such as SRT or simpler ones such as RTMP or DASH.

 

 

Zixi

The ZIXI protocol emerged as an advanced solution for the contribution and distribution of content in broadcast environments over the Internet. Developed by ZIXI, this protocol focuses on ensuring reliable and low-latency delivery without compromising the quality of the content transmitted by adapting to the situation of the network at all times.

One of the key features of ZIXI is its bandwidth management capability. By using smart algorithms, the protocol can dynamically adapt transmission rate and compression of the video stream based on network conditions. This ensures optimal use of available bandwidth and enables high-quality transmission even in environments with fluctuations in bandwidth.

In addition, it features advanced error correction and selective retransmission mechanisms to ensure the integrity of the transmitted data. This allows the protocol to recover lost or damaged packets in real time, minimizing the effects of packet loss on the quality and consistency of the transmitted content.

Low latency is another crucial aspect in real-time broadcast transmission and the ZIXI protocol addresses this concern effectively. The protocol has been designed to minimize latency in video and audio transmission, enabling real-time collaboration and interaction between different contribution and distribution points. This is especially valuable at live events, where accurate timing is essential.

It also offers a complete solution for the security of the transmission. This protocol supports end-to-end encryption, thus ensuring the confidentiality and integrity of the transmitted data. In addition, it includes authentication and access control mechanisms to protect against unauthorized access and ensure the authenticity of participants in the transmission.

By its nature, ZIXI is primarily used in distribution environments or as a contribution between live events and production sites. In fact, there are many manufacturers that have implemented their own protocols based on ZIXI, since the latter can serve as the basis for other more advanced implementations. It has been gradually replaced by more modern protocols such as SRT, but there are still many environments in which ZIXI is a key protocol.

 

Dynamic Adaptive Streaming over HTTP (DASH)

The DASH protocol is based on the HTTP standard and uses adaptive streaming techniques to deliver multimedia content efficiently. Instead of streaming a single video stream, DASH splits content into smaller segments and delivers them to the receiving device over standard HTTP connections. These segments can have different resolutions, bitrates, and compression formats.

One of the main advantages of DASH is its ability to adapt dynamically. The protocol constantly monitors network conditions and receiver device capabilities, and then adjusts video quality in real time to ensure smooth and seamless playback. Thus, the content automatically adapts to the network connection and the capabilities of the device, optimizing the viewing experience for each user.

Another key feature of DASH is its support for multiple video formats, such as MPEG-DASH, HLS and Smooth Streaming. This enables content providers to tailor their broadcasts to different devices and platforms, thus ensuring broad accessibility and a consistent user experience across a variety of devices, from smart TVs to mobile phones.

And of course, this is 100% aimed at distributing content to end users via the Internet. Obviously it can also be used in more controlled IPTV networks, but its potential is fully realized when content has to be delivered to millions of end users having different devices each. Obviously, DASH relies on CDN networks for such distribution.

In addition, this standard offers significant benefits in terms of storage and scalability. Content segments are cached on the client side, thus reducing the load on streaming servers and facilitating efficient delivery to a large number of concurrent users. This makes the DASH standard a cost-effective option for mass distribution of online content.

The DASH protocol also allows users to pause, fast-forward, or rewind content seamlessly. Thanks to segment-based delivery, users can jump to any point of the content without the need to download the entire video file whenever the origin of the CDN allows. This provides greater flexibility and comfort for viewers, enhancing their viewing experience.

 

HTTP Live Streaming (HLS)

The HLS standard uses the HTTP protocol to stream multimedia content in small, adaptive segments, in similar way to DASH. Content is divided into fixed-duration segments; these are stored on source servers and delivered to the receiving devices over standard HTTP connections via CDN networks. These segments can have different resolutions, bitrates, and compression formats. This allows for dynamic adaptation based on network conditions and device capabilities.

One of the key advantages of HLS is its compatibility with a wide range of devices and platforms. This protocol supports web browsers, mobile operating systems and streaming devices, ensuring users can access content seamlessly from a variety of devices, whether mobile phones, tablets, smart TVs or computers.

HLS also offers an optimized viewing experience through adaptive streaming. The protocol constantly monitors network conditions and receiving device capabilities, and then adjusts the most suitable video quality in real time to ensure smooth and seamless playback. This means that users will experience optimal video quality based on their network connection and the capabilities of their device.

Another important advantage of HLS is its ability to handle interruptions and fluctuations in the network connection. If a momentary outage or decrease in network quality occurs, this standard can be seamlessly switch to lower video quality to avoid interruptions in playback. This ensures that viewers can enjoy the content without trouble, even on networks with bandwidth limitations.

In the same way as DASH, HLS is completely oriented to distribution of content over the Internet, OTT and IPTV environments. Together with DASH they are the most widespread protocols and, I would venture to say that, due to its very high compatibility with almost any device, HLS is the most popular protocol on the Internet for distribution of streaming content.

Furthermore, HLS offers benefits in terms of storage and transmission of content on demand. Content segments are cached on the client side, thus reducing the load on streaming servers and facilitating delivery to a large number of concurrent users. This makes the HSL standard a a scalable, profitable option for mass distribution of online content.

 

Real-Time Messaging Protocol (RTMP)

RTMP was developed by Macromedia (now Adobe) and has become a de-facto standard for live streaming and real-time interaction on the Internet. This protocol uses a persistent connection between the transmitting server and the receiving client, allowing for continuous and reliable transmission of the multimedia content.

One of the main advantages of RTMP is its low latency. The protocol has been designed to minimize the delay between streaming and playback of content, making it ideal for live applications where accurate synchronization is essential. This allows viewers to enjoy real-time events, such as sports broadcasts or concerts, without significant delays.

In addition, the RTMP standard offers high-quality media transmission. The protocol enables content delivery in different formats and bitrates, ensuring smooth playback and optimal video and audio quality. This is especially important in broadcast environments, where content quality is paramount.

RTMP is also known for its ability to adapt to network conditions. It can dynamically adjust the transmission quality based on the available bandwidth and fluctuations in the connection. This ensures stable and uninterrupted transmission, even on networks with bandwidth limitations or unreliable links.

In terms of application in broadcast contribution and distribution environments, it offers flexibility and compatibility with a variety of devices and platforms. Content providers can use RTMP to stream video and audio signals from different sources such as studio cameras, mobile production units or live streaming applications, to streaming and distribution servers.

The RTMP protocol is also widely used in broadcast distribution through online platforms and streaming services. It allows content providers to deliver live streams or prerecorded content to a wide audience, thus providing an interactive, real-time viewing experience.

Although RTMP has been widely used in the past, it is important to mention that its popularity has declined in recent years due to the emergence of more modern and efficient protocols, such as HLS and DASH. However, it remains relevant in certain use cases and specific environments that require low latency and real-time transmission.

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