Adam K, Author at The Broadcast Knowledge

Video: RIST and Open Broadcast Systems

Posted on 6th May 2020 by Adam K

RIST is a streaming protocol which allows lossy networks such as the internet to be used for critical streaming applications. Called Reliable Internet Stream Transport, it uses ARQ (Automatic Repeat reQuest) retransmission technology to request any data that is lost by the network, creating reliable paths for video contribution.

In this presentation, Kieran Kunhya from Open Broadcast Systems explains why his company has chosen RIST protocol for their software-based encoders and decoders. Their initial solution for news, sports and linear channels contribution over public internet were based on FEC (Forward Error Correction), a technique used for controlling errors in transmission by sending data in a redundant way using error-correcting code. However, FEC couldn’t cope with large burst losses, there was limited interoperability and the implementation was complex. Protecting the stream by sending the same feed over multiple paths and/or sending a delayed version of the stream on the same path, had a heavy bandwidth penalty. This prompted them, instead, to implement an ARQ technique based on RFC 4585 (Extended RTP Profile for Real-time Transport Control Protocol-Based Feedback), which gave them functionality quite similar to the basic RIST functionality.

Key to the discussion, Kieran explains why they decided not to adopt the SRT protocol. As SRT is based file transfer protocol, it’s difficult or impossible to add features like bonding, multi-network and multi-point support which were available in RIST from day one. Moreover, RIST has a large IETF heritage from other industries and is vendor-independent. In Kieran’s opinion, SRT will become a prosumer solution (similar to RTMP, now, for streaming) and RIST will be the professional solution (analogous to MPEG-2 Transport Streams).

Different applications for the RIST protocol are discussed, including 24/7 linear channels for satellite uplink from playout, interactive (two-way) talking heads for news, high bitrate live events and reverse vision lines for monitoring purposes. Also, there is a big potential for using RIST in cloud solutions for live broadcast production workflows. Kieran hopes that more broadcasters will start using spin-up and spin-down cloud workflows, which will help save space and money on infrastructure.

What’s interesting, Open Broadcast Solutions are not currently interested in RIST Main Profile (the main advantages of this profile are support for encryption, authentication and in-band data). Kieran explains that to control devices in remote locations you need some kind of off-the-shelf VPN anyway. These systems provide encryption and NAT port traversal, so the problem is solved at a different layer in the OSI model and this gives customers more control over the type of encryption they want.

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Speaker

Kieran Kunhya
Founder and CEO,
Open Broadcast Systems

Video: RIST in the Cloud

Posted on 22nd April 2020 by Adam K

Cloud workflows are starting to become an integral part of broadcasters’ live production. However, the quality of video is often not sufficient for high-end broadcast applications where cloud infrastructure providers such as Google, Oracle or AWS are accessed through the public Internet or leased lines.

A number of protocols based on ARQ (Adaptive Repeat reQuest) retransmission technology have been created (including SRT, Zixi, VideoFlow and RIST) to solve the challenge of moving professional media over the Internet which is fraught with dropped packets and unwanted delays. Protocols such as a SRT and RIST enable broadcast-grade video delivery at a much lower cost than fibre or satellite links.

The RIST (Reliable Internet Streaming Transport) protocol has been created as an open alternative to commercial options such as Zixi. This protocol is a merging of technologies from around the industry built upon current standards in IETF RFCs, providing an open, interoperable and technically robust solution for low-latency live video over unmanaged networks.

In this presentation David Griggs from Amazon Web Services (AWS) talks about how the RIST protocol with cloud technology is transforming broadcast content distribution. He explains that delivery of live content is essential for the broadcasters and they look for a way to deliver this content without using expensive private fibre optics or satellite links. With unmanaged networks you can get content from one side of the world to the other with very little investment in time and infrastructure, but it is only possible with protocols based on ARQ like RIST.

Next, David discusses the major advantages of cloud technology, being dynamic and flexible. Historically dimensioning the entire production environment for peak utilisation was financially challenging. Now it is possible to dimension it for average use, while leveraging cloud resources for peak usage, providing a more elastic cost model. Moreover, the cloud is a good place to innovate and to experiment because the barrier to entry in terms of cost is low. It encourages both customers and vendors to experiment and to be innovative and ultimately build more compelling and better solutions.

David believes that open and interoperable QoS protocols like RIST will be instrumental in building complex distribution networks in the cloud. He hopes that AWS by working together with Net Insight, Zixi and Cobalt Digital can start to build innovative and interoperable cloud solutions for live sports.

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Speaker

David Griggs
Senior Product Manager, Media Services
AWS Elemental

Video: Encoding and packaging for DVB-I services

Posted on 2nd April 2020 by Adam K

There are many ways of achieving a hybrid of OTT-delivered and broadcast-delivered content, but they are not necessarily interoperable. DVB aims to solve the interoperability issue, along with the problem of service discovery with DVB-I. This specification was developed to bring linear TV over the internet up to the standard of traditional broadcast in terms of both video quality and user experience.

DVB-I supports any device with a suitable internet connection and media player, including TV sets, smartphones, tablets and media streaming devices. The medium itself can still be satellite, cable or DTT, but services are encapsulated in IP. Where both broadband and broadcast connections are available, devices can present an integrated list of services and content, combining both streamed and broadcast services.

DVB-I standard relies on three components developed separately within DVB: the low latency operation, multicast streaming and advanced service discovery. In this webinar, Rufael Mekuria from Unified Streaming focuses on low latency distributed workflow for encoding and packaging.

The process starts with an ABR (adaptive bit rate) encoder responsible for producing streams with multiple bit rates and clear segmentation – this allows clients to automatically choose the best video quality depending on available bandwidth. Next step is packaging where streaming manifests are added and content encryption is applied, then data is distributed through origin servers and CDNs.

Rufael explains that low latency mode is based on an enhancement to the DVB-DASH streaming specification known as DVB Bluebook A168. This incorporates the chunked transfer encoding of the MPEG CMAF (Common Media Application Format), developed to enable co-existence between the two principle flavors of adaptive bit rate streaming: HLS and DASH. Chunked transfer encoding is a compromise between segment size and encoding efficiency (shorter segments make it harder for encoders to work efficiently). The encoder splits the segments into groups of frames none of which requires a frame from a later group to enable decoding. The DASH packager then puts each group of frames into a CMAF chunk and pushes it to the CDN. DVB claims this approach can cut end-to-end stream latency from a typical 20-30 seconds down to 3-4 seconds.

The other topics covered are: encryption (exhanging key parameters using CPIX), content insertion, metadata, supplemental descriptors, TTML subitles and MPD proxy.

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Download the slides.

Speaker

Rufael Mekuria
Head of Research & Standardization
Unified Streaming

Video: Reinventing Intercom with SMPTE ST 2110-30

Posted on 18th March 2020 by Adam K

Intercom systems form the backbone of any broadcast production environment. There have been great strides made in the advancement of these systems, and matrix intercoms are very mature solution now, with partylines, IFBs and groups, wide range of connectivity options and easy signal monitoring. However, they have flaws as well. Initial cost is high and there’s lack of flexibility as system size is limited by the matrix port count. It is possible to trunk multiple frames, but it is difficult, expensive and takes rack space. Moreover, everything cables back to a central matrix which might be a single point of failure.

In this presentation, Martin Dyster from The Telos Alliance looks at the parallels between the emergence of Audio over IP (AoIP) standards and the development of products in the intercom market. First a short history of Audio over IP protocols is shown, including Telos Livewire (2003), Audinate Dante (2006), Wheatstone WheatNet (2008) and ALC Networks Ravenna (2010). With all these protocols available a question of interoperability has arisen – if you try to connect equipment using two different AoIP protocols it simply won’t work.

In 2010 The Audio Engineering Society formed the x192 Working Group which was the driving force behind the AES67. This standard was ratified in 2013 and allowed interconnecting audio equipment from different vendors. In 2017 SMPTE adapted AES67 as the audio format for ST 2110 standard.

Audio over IP replaces the idea of connecting all devices “point-to-point” with multicast IP flows – all devices are connected via a common fabric and all audio routes are simply messages that go from one device to another. Martin explains how Telos were inspired by this approach to move away from the matrix based intercoms and create a distributed system, in which there is no central core and DSP processing is built in intercom panels. Each panel contains audio mix engines and a set of AES67 receivers and transmitters which use multicast IP flows. Any ST 2110-30 / AES67 compatible devices present on the network can connect with intercom panels without an external interface. Analog and other baseband audio needs to be converted to ST 2110-30 / AES67.

Martin finishes his presentation by highlighting advantages of AoIP intercom systems, including lower entry and maintenance cost, easy expansion (multi studio or even multi site) and resilient operation (no single point of failure). Moreover, adaptation of multicast IP audio flows removes the need for DAs, patch bays and centralised routers, which reduces cabling and saves rack space.

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Download the slides.

If you want to refresh your knowledge about AES67 and ST2110-30, we recomend the Video: Deep Dive into SMPTE ST 2110-30, 31 & AES 67 Audio presentation by Leigh Whitcomb.

Speaker