NETGEAR will showcase its new switch models and major updates to its Engage Control software during the 2026 NAB Show, April 18-22, at the Las Vegas Convention Center.

The company has expanded its M4350 switch portfolio with the addition of the new M4350-16M4V and M4350-16C.

The M4350-16M4V offers secure, lockable connectivity with Neutrik etherCON, opticalCON QUAD and powerCON TRUE1 connectors. It features 16x 2.5G PoE++ ports, including eight etherCON, plus 4x 25G SFP28 uplinks via a modular card slot Optional uplink cards include RJ-45 or opticalCON QUAD connectors for single-mode or multimode fiber.

The new M4350-16C model addresses throughput challenges in large AV-over-IP deployments. With 16 ports of 100G connectivity, the M4350-16C delivers the bandwidth required for aggregation and core layers where multiple high-resolution video streams converge.

The M4350 series offers enterprise-grade managed switching from 1G to 100G built for AV and IT networks. With high-power PoE++, redundant modular power and hitless failover, it ensures mission-critical reliability for broadcast and AV-over-IP.

The new models support SMPTE ST 2110 timing with grandmaster and boundary clock functionality. Trade Agreement Act (TAA)-compliant SKUs are available for government use.

NETGEAR will also have its network design services team at the NAB Show to answer the questions of attendees. The company will also give personalized demos of its updated Engage Controller version 2.4, which features NETGEAR’s profile-based approach for reliable, simplified switch configuration.

NETGEAR will also offer on-demand sessions at its booth on the design of IP-based broadcast networks for live production and facility environments. More than 110 NETGEAR partner manufacturers will show their broadcast solutions powered by the company’s networking technology.

See NETGEAR at 2026 NAB Show booth C.7303.

More information is available on the company’s website.

MARINA DEL RAY, Calif.—Last week, the Video Services Forum (VSF)—an industry group covering all aspects of video networking technology—celebrated the 25th anniversary of VidTrans, its annual three-day gathering. Usually held the week after the HPA Tech Retreat in Palm Springs, the timing and proximity makes it easier for Tech Retreat attendees to  participate in both events.  

This year, 148 industry professionals attended the event, Feb. 28-March 2, which despite the not-so-friendly southern California (wet/cold) weather, went on without a hitch. 

Topics of discussion covered the transport of video and media signals throughout systems, hence the name “VidTrans.”  This year’s agenda focused on innovative types of networking and video technologies along with their application to video transport. 

Here’s a summary of selected presentations:

Video Quality Monitoring
In the session “OTT: Best Practices and Challenges,” Andrey Pozdnyakov, president of  video monitoring provider Elecard, discussed typical OTT tasks and challenges, practical use-cases and quality monitoring models. Andrey described how to detect problematic parts of technical equipment or networks which then quickly identify who is responsible for the issue. 

Elecard’s monitoring products describe the differences and findings in a Quality of Services (QoS) vs. a Quality of Experience (QoE), similar to those applications in cable or encoding systems. 

Implementing Sustainability
In a hybrid session that included in-person and virtual panelists, “Minimizing the Carbon Footprint of Live Production” focused on sustainability in the video networking space. 

Moderated by Andy Rayner, chief technologist at Nevion, panelists Thomas Edwards of AWS, Barbara Lange of Kibo121, Athena Trastelis of the CBC, and Geoff Wolf of the BBC, discussed how their organizations are working towards implementing and maintaining a carbon neutral-future, focusing on the environmental impacts and best practices in the workplace.  Panelists asked the VidTrans audience to consider buying products from companies who have a demonstrable plan for improving their workplace or products amidst a growing global awareness for sustainability.

Live Cloud Production Networking
On-demand networking for enabling live cloud sports production in a real-world proof of concept (PoC) was discussed by Brad Cheney of Fox Sports, Thomas Edwards of AWS, and Ryan Korte of Cloud Connect, Lumen Technologies. The team showed how they approached “working from home” for live sports productions including the products and practices, showing examples of recent live sports productions done entirely in the cloud from non-venue locations.

Prompted by the impact of Covid on remote production, Fox in particular, has taken a stunning direction in making live production possible using cloud-based services and production support services that literally “sit” in production personnels’ living rooms, front porches or in a hotel.

This is obviously changing the structure, feasibility, and cost of doing live sports productions by mitigating the expenses and time associated with moving an entire truck fleet-level staff to the onsite venue. This new approach is changing the model by spreading the production into geographically separated locations and eliminating the requirements to transport large sets of complex hardware (i.e., mobile outside broadcast units) to site.

Designing for Efficiency
Adam Salkin, director of engineering, M&E West for Diversified, discussed how EVS, TAG VS and Fox Sports collaborated with Diversified to design, assemble and configure a fully transportable system (Fig. 1a) contained in a system engineered for transport in a cargo plane’s belly.

The folding rack system (Fig. 1b) consists of multiple skids that hold five 44RU systems each and contain a complete ST 2110 mobile production system that was used for Fox Sports’ coverage of the 2022 FIFA World Cup, Super Bowl LVII and the upcoming 2023 FIFA Women’s Cup this summer in Australia and New Zealand.

Considering the complexities associated with assembling and transporting the huge volume of gear needed to technically produce these events, it was impractical to use ocean cargo methods to get the systems between locations in the time frame available between two major events. Developing the air-cargo system was essential to covering the World Cup and Super Bowl in the minimal time allotted.

‘Building the Airplane While in Flight’
I had the pleasure of joining Microsoft’s technology team of Jack Clawson, Chief Solutions Architect and John Ball, Project Manager at Microsoft to focus on the timelines involved in transitioning to IP in the session “End User Requirements and Design Challenges,” moderated by Brad Gilmer, Gilmer and Associates. 

In the session, Jack and John outlined all the detailed planning and implementation requirements for an active project that updated Microsoft’s Production Studios services on their Redmond, Wash. Campus.  

Panelists talked about the steps taken in the migration from an SDI-based system to IP. Since the five-year-old SMPTE ST2110 standard is so new, such a transition was described by one panelist as akin to “building the airplane while it’s in flight.”  

Discussion topics focused on how the integration addressed supply-chain issues, delays, and the continual changes and additions incurred as the design and install was developed throughout an 18-month period; all the while, keeping live production going and major events Microsoft conference events moving forward. Extensive adjustments were constantly made as switch deliveries and new products were introduced into the Microsoft Production Studio systems—often with delay adjustments driving the “next phase” of each project segment.

Network Details
Chris Lapp, Technical Solutions Architect of Cisco Systems discussed any-source multicasting (ASM) in the session “PIM: there's a better way,” which included a brief overview of PIM-DM (dense mode) and PIM-SM (sparse mode). In addition, Chris described the concepts of “PIM flooding,” a generic way of distributing information within a PIM domain as well as “PIM source discovery,” which announces information of active sources throughout a PIM domain. 

Chris discussed the status of this technology within the IETF and Cisco, describing how the oldest multicast routing protocol commonly used today is defined in RFC 3973 (which is only an experimental protocol and not a standard). 

Sufficient detail was diagrammed to describe the variations in the family of PIM (protocol independent multicast) as optimized for specific, particular, or differing environments. The protocols provide one-to-many and many-to-many distribution of data over a LAN, WAN, or the Internet.  

IPMX and RIST Updates
One of VidTrans’s annual rites of passage include a summary status of the variations, findings, and demonstrations of the newest Technical Recommendations, such as TR-10 for Internet Protocol Media eXperience (IPMX) and TR-06 Reliable Internet Stream Protocol (RIST). 

The developers for both of these transport technology TRs discussed their progress within the VSF Activity Group, including an update on TR-10 within the IPMX group from Jack Douglas, Vice President Marketing and Business Development for Packetstorm. IPMX is a ST 2110-type implementation focused on products in the ProAV marketplace. Such products are already available for harmonized implementation with ST 2110 systems.

Dr. Ciro A. Noronha, Ph.D, Cobalt Digital, updated the gathering on the latest developments in RIST Source Adaptation. Topics included the motivation for source adaptation, and an overview of the VSF TR-06-4 (Part 1) and preliminary implementation results for this technology. 

TR-06-4 (part 1) is designed to recover packet loss via retransmission. When network capacity falls below the stream rate, no amount of retransmission can fix this (i.e., you can’t recover a 5 Mbps stream once the link falls below 4 Mbps). TR-06-4 sets a point where the stream source must be reduced to compensate for a link level that is insufficient. This TR creates a signaling methodology to dynamically adjust the stream source rate to compensate for network changes.

New features are summarized in TR-06-3 RIST Advanced Profile for other protocols and in TR-06-4 RIST Ancillary Features including Source Adaptation (approved in November 2022) and the use of Wiregaurd VPN in RIST (approved in January). 

Historical Perspectives and PTP Security
Wes Simpson of LearnIPVideo reviewed the history of the Video Services Forum, from its creation in 1997 to the present, while John R. Naylor, Vice President of DashBoard, softGear at Ross Video, presented the SMPTE Study Group’s second report on Security in ST 2059, including a detailed overview of how PTP (i.e., IEEE-1588) functions, typical PTP applications, how PTP is used in broadcast infrastructures and the vulnerabilities of PTP on systems plus suggested corrective actions to prevent bad actors from taking over PTP in a system.

The VSF’s annual technical conference is an educational and networking experience for all those working in the domain of IP video in a broadcast environment, a WAN or a transport solution provider’s world.  Each year the VidTrans conference showcases some of the latest applications, technologies and products in its exhibit hall as well as provides attendees with networking opportunities at this special event. 

More information about VSF is available at vsf.tv.

Editors note: Updated, Nov. 7.

WASHINGTON--Government Video Expo, the Mid-Atlantic’s largest video trade show, is partnering with SMPTE for an all-day series of tutorials on the basics of IP for media, including a look at SMPTE-2110, the new standard for video transport over IP. 

“SMPTE in DC: Essential Technology Advances for Media Pros” will take place Wednesday, Nov. 28., and you can register to attend here. For more information on the show, visit www.gvexpo.com. 

Here is the agenda:

9–9:30 a.m.

• Open, Welcome and an Introduction to Professional Video Over IP

Video over IP networks and the internet has existed for more than two decades. Why is SMPTE developing Studio Video over IP (SVIP) — a new set of standards for video over IP? Why are they needed? What is the difference between streaming video and SVIP?

Peter Wharton, President, Happy Robotz

9:30-10:30 

• High Dynamic Range: The Best TV Picture You’ve Ever Seen

This HDR presentation will introduce HDR and discuss HDR standards, terminology and the history of dynamic range. The advanced part of the presentation will cover HDR transfer curves and comparing HDR profiles, as well as compatibility with SDR displays and the consumer adoption of UHD HDR TVs. Although Ultra HD and HDR are usually mentioned together, high-dynamic range is not limited to UHD. HDTV can also benefit from a dramatic improvement in picture quality. HDR offers a more realistic picture, similar to the way we see things in real life. In the future, it’s likely that multiple HDR profiles, optimized for either live TV or post-produced movies, will be utilized.

John Humphrey, Vice President Business Development, Hitachi

11:00-Noon

• Building Scalable Facilities Using SMPTE 2110

The new set of SMPTE standards (ST 2110) provides a technical underpinning for building large-scale media facilities with uncompromised picture quality and low latency, with scalability to large image formats (4K, 8K and beyond) and advanced color depths and spaces. This talk is an overview of the new standards, and also covers the practical details of implementing large-scale systems including control APIs and network topology.

John Mailhot, Systems Architect for IP Convergence, Imagine Communications

Noon–1:00 p.m.

• How Smooth Are Your Packets? Implementation Realities and Best Practices of IP and PTP

This session is a tutorial on the what you need to know to understand the challenges we face in the process of making the transition to IP-based transport for video, audio and data. This move toward infrastructure efficiency has brought new technical challenges requiring broadcast engineers to gain an understanding of the technology and the latest techniques needed to monitor these signals.

The development of SMPTE ST 2110 is a suite of standards that provide encapsulation of uncompressed video within IP packets and for live IP production carrying separate streams of video, audio and data packets. This new standard also allows for SMPTE ST 2022-6 that provides encapsulation of uncompressed SDI as well as ST 2059 for system timing.

This session will take a look at the basic structure of the packets for ST 2022 and ST 2110, how variable delay across the network introduces jitter at the receiver and how measurements can be made on the stream. Latency in the network can produce out-of-order packets or corruption of the data causing packets to be dropped. Therefore, it is important to monitor the stream to ensure an error-free network to ensure transmission of the high bit rate media and how these errors affect the actual video and audio signal. For redundancy in the media network, SMPTE ST 2022-7 can be used to provide a Path One and a Path Two stream so that the downstream device can determine which path is the most appropriate to use. Measurement of the integrity of both paths is important and we will look at ways of monitoring the signal paths.

Karl Kuhn, Sr. Field Video Application Engineer, Tektronix

1–2 p.m.
LUNCH

2:00–3:00

Myths and Realities of Security of Professional Media Over Managed IP Networks

It seems that nearly every day there is another story about an IP network breach. Now "they" want to build professional media creation facilities with IP at the core? What are the risks? And, are the risks as bad as "they" say? Find out the facts about IP network security in this must-see SMPTE session at Government Video Expo!

Thomas Bause Mason, Director Standards Development, SMPTE

3:00–4:00

• The People Side: Managing Change and Skills Transition

The accelerating pace of technological change requires an increased focus on an organization's top resource — its people. This presentation will provide insights, practical approaches and tools for managing change, building and retaining a diverse and inclusive workforce.

John McCoskey, Industry Executive, Eagle Hill Consulting - Technology, Media & Entertainment

For more information and to register for GV Expo, visit www.gvexpo.com.

Technology-related content delivery challenges are taking a back seat to business issues as the online video segment of the industry matures. New research conducted by Broadcasting & Cable and TV Technology magazines in partnership with Comcast technology Solutions, confirms this shift, and many others as content and video providers (CAVPs) shared their strategies for optimizing the business and technical performance of their video distribution services.

This report lays out these details including:

• Controlling costs and maximizing customer lifetime value are important, but not the top business challenge
• 3 key features CAVPs identified as differentiators to stay ahead of their competition
• Why half of the responders would put most of their existing service in the cloud
• How clip sharing can be a critical component to building subscriber numbers

Click here to download the full white paper.

This is the third installment in a series of articles about the newly-published SMPTE standard covering elementary media flows over managed IP networks. This month, the focus is again on video transport, specifically the rules that help ensure that high-bitrate video streams are well-behaved and won’t overwhelm the IP networks used to transport them nor overflow receiver buffers.

KEEPING STREAMS FROM OVERFLOWING

The full title of SMPTE ST 2110-21 is “Professional Media Over Managed IP Networks: Traffic Shaping and Delivery Timing for Video.” Both of these terms refer to the same basic topic: how are the packets transmitted over the network, from the perspective of both the sender and the receiver? In other words, how should packet flows be sent into the network “pipes” so as to not cause flooding (of packets)? Answering these questions is crucial for properly provisioning network connections to support as many signals as possible without causing packet congestion, which could lead to packet loss.

To really understand the potential issue, it helps to deal with some actual numbers. Consider a 1080p signal with a 50Hz (European) frame rate that has a bandwidth of 3 Gbps on an SDI cable. At 50 Hz, a new frame of video is created every 20 milliseconds. Using constant-size packets, each with 480 pixels, would result in 4320 packets for each video frame. Using 10-bit sampling, 480 pixels would require 1200 bytes of data, plus 90 bytes of overhead, for a total of 1290 bytes per packet.

One way that this video signal could be transmitted would be using evenly spaced packets. In this case, 4320 packets sent every 20 milliseconds would mean that one packet is sent every 4.63 microseconds. On a 10 Gbps link, each 1290-byte packet would require (1290*8=10,320) bit-times to transmit, or 1.032 microseconds, followed by a gap of roughly 3.6 microseconds.

Another way that this video signal could be sent would be sending all of the pixels within a frame all at once. (This could hypothetically be the case for a device that was, say, generating graphics.) In this case, the sender would generate a burst of packets lasting 4320*10,230 bit times, or 4.419 milliseconds. This burst of packets would be followed by a gap of (20-4.419) 15.581 milliseconds until the next video frame was ready to transmit.

[SMPTE ST 2110-10: A Base to Build On]

So which of these packet transmission schemes is better? Well, both of the above streams have the same long-term average bit rate, of just about 2.23 Gbps. However, the first stream with evenly spaced packets will be much easier for signal receivers, switches and other network devices to handle. Why? Because the second stream will fully occupy a 10 Gbps data circuit for a significant period of time, forcing any lower-priority data that might need to be transmitted over that link to be placed in a buffer to wait for the link to become free. If higher-priority data came along during the data burst, then video packets would need to be buffered instead. Since network devices typically have a limited amount of buffer space that needs to be shared by multiple physical data channels, any signal that places heavy demands on buffers can cause problems, such as lost or deleted packets when buffers are filled up.

This issue becomes even more of a problem when multiple senders are all trying to transmit bursts of data at the same time, as would frequently occur in applications where several video sources are locked to a common clock. Therefore, to prevent network problems and to make it easier to design signal receivers, it makes sense to set some limits on the size and duration of packet bursts. These limits are often called “traffic shaping” and/or “delivery timing” in networking jargon.

TYPE N, NL AND W VIDEO SOURCES

The ST 2110-21 standard defines three types of senders: N (for Narrow), NL (for Narrow Linear) and W (for Wide). These types define limits for the amount of packet delay variation (i.e. the burstiness) that a sender is allowed to exhibit in its output stream.

Type NL is the easiest to understand, and corresponds to a stream where all the packets for a video signal are evenly spaced across the duration of each video frame (i.e. a stream that is like the first example given in the preceding section). The SDP (Session Description Protocol) for this type of stream must include a parameter “TP=2110TPNL.”

[SMPTE ST 2110-20: Pass the Pixels, Please]

Type N is similar to Type NL, except that the sender doesn’t send packets during the time that would correspond to the VBI (Vertical Blanking Interval) or VANC (Vertical Ancillary data space) of the corresponding traditional SDI video signal. Thus, a Type N sender would be able to send packets in a stream that would have a noticeable gap that occurs during each video frame period. For example, in a 720p signal running at 50 frames per second, and a VANC equal in duration to 30 lines of video, the sender would deliver packets for (720/750*20 milliseconds) 19.2 milliseconds out of every 20 milliseconds, and have a 0.8 millisecond gap when no packets are sent. Note that this would be the behavior that would be the easiest to implement if an incoming SDI signal was simply converted to ST2110 packets whenever active video samples arrive. The SDP for this type of stream must include a parameter “TP=2110TPN.”

Type W senders are allowed to have a significantly greater burstiness. This category was included in the ST 2110-21 standard to accommodate software-based senders, such as a graphics generator. For Type W flows, senders can have at least quadruple the amount of burstiness as a Type N or a Type NL, and in many cases much more. While this looser tolerance should make things better for a sender (particularly one that is implemented on a virtual machine), it does have consequences for receivers, which require a corresponding increase in the size of their input buffers. This can be costly, both in terms of the raw amount of memory that is required as well as in terms of delay for an incoming signal. Type W streams will also tend to consume larger amounts of buffer space within network switches and other devices; applications with significant quantities of Type W senders will need to be implemented using network devices that have sufficiently large internal memories. The SDP for this type of stream must include a parameter “TP=2110TPW.”

Fig. 1

Click on the Image to Enlarge

Fig. 1 shows a comparison of the three different sender types. Each of the three types is shown with a simplified representation of the packet flows along with a moving average of the bit rate. Note that for type NL, the moving average bit rate is flat, showing that these flows are the best behaved. Type N shows an increase in average bit rate during active video and a decline during the VANC. For Type W, the average rate shows even larger flow rate surges and drops.

Receivers are also categorized in ST 2110-21, but this information is not required to be transmitted with SDP. A Type N receiver should be able to correctly receive a flow originating from either a Type N or a Type NL sender(but not a Type W), provided that the receiver is locked to the same clock as the sender and that sender is aligned to the SMPTE ST 2059-1 Epoch. A Type W receiver should be able to receive a stream from an N, NL or W sender, provided the receiver is locked to the same clock as the sender. A Type A (for Asynchronous) receiver should be capable of receiving a stream from any type of sender, regardless of clock source or signal phase.

PRACTICAL CONSIDERATIONS

For time-sensitive flows within a media facility, such as a live broadcast, Type N or NL senders will tend to dominate, because they are the easiest to multiplex together and require the least amount of buffering in the network and at receivers, and therefore introduce the smallest amount of end-to-end delay. If Type W senders are present in a network, receivers will also need to be Type W to be able to receive these flows. For non-frame-accurate applications, such as monitors and multiviewers, or when synchronization between senders and receivers is not present, Type A receivers can be used to accommodate any type of ST 2110 uncompressed video flow.

Wes Simpson is the president of Telecom Product Consulting. He can be reached via TV Technology.

The verdict is in: 5G wireless systems will be all things to all people, allowing incredibly high bandwidths and ultra-low power radios to be used anywhere, all at a fraction of the cost of today’s devices. Industry journals are filled with reports of successful deployments, and consumers are getting ready for the next generation of smartphones to transport them to a future of connectivity nirvana.

Does this sound too good to be true? Well it is, at least from where the wireless industry stands in 2017. Because the final standards have yet to be ratified, there are lots of interesting trials and experiments underway to move towards a common standard that will be required before handset manufacturers and mobile system operators are able to begin mass deployments. But, as things start to come into focus, it is readily apparent that 5G will transform wireless networking as we know it.

WHAT IS 5G?
Each of the past generations of wireless networking has been named in a series, starting with 1G, which were the first cellphones that used analog channels to provide voice communications. 2G introduced digital radios and text messaging; one popular version was named “GSM.” 3G increased the data channel capacities to at least 200 Kbps, with enhancements that reached into the Megabit range. 4G and 4GLTE are the most advanced standards in use today, and provide all of the features and functions that modern smartphones utilize for delivering streaming video, multimedia messages and millions of voice connections.

The ITU (International Telecommunication Union) is set to ratify a set of minimum requirements (called ITU-R M.IMT-2020 for 5G systems during their November 2017 meeting. These requirements include:

20 Gbps peak downlink rate to a single mobile station, in a perfect environment with all available resources utilized;

10 Gbps peak uplink rate, also under ideal conditions;

100 Mbps user experienced downlink rate, which is the worst data rate that users who are within range of the network should encounter ninety-five percent of the time. This is roughly a hundred times what many current mobile networks are designed to deliver;

50 Mbps user experienced uplink rate, which will be important for wireless cameras and other applications;

4 msec one-way latency across the radio link for enhanced mobile device users, which will help improve throughput for streaming video and file uploads/downloads, and

Up to 1 million simultaneous connections per square kilometer, which means that supporting a dense collection of wireless, low power Internet of Things (IoT) devices will actually be feasible.

One of the ways that 5G will attain all of these impressive performance numbers is by using radios that are capable of delivering 30 bits per second per Hz peak spectral efficiency downlink and 15 bps/Hz uplink, which requires advanced modulation schemes and antennas to be coupled with powerful digital signal processing. Improvements in base station density (number of base stations per unit area) and use of millimeter-wave (30-300 GHz) signals will also be necessary to achieve some of 5G’s performance targets.

IMPACT ON BROADCASTERS
The advances brought by 5G technology are already being experienced by a few lucky users who are involved in early trials of fixed broadband connections based on millimeter-wave radio technologies. These signals, which require high-gain antennas working over direct line-of-sight links, are not able to pass through objects and can experience significant fading due to rain. In spite of these limitations, 5G fixed wireless technology could make it easier (and less expensive) for carriers to provide high-bandwidth connections to broadcasters who have facilities in so-called “digital deserts” and bring more competition into the local loop market as early as 2018 in some cities.

Broadcasters are increasingly using portable cameras with wireless backhaul for news, sports and remote event coverage, and the advances brought by 5G technology should have a major impact on performance. Unfortunately, broadcasters’ ability to use these systems will be delayed until carriers are able to deploy base stations that support these technologies in the areas where broadcasters will want to use them. Mass deployment will have to wait for industry standards to be finalized.

Speaking of standards, work is underway in several different committees to define exactly which technologies will be selected to implement 5G’s lofty performance goals. Of course, these decisions will need to balance the needs of users and carriers, patent holders and manufacturers, and result in a system that can be economically implemented using technologies that are going to be available within the next year or two. To help accelerate the standards process, the 3GPP (Third Generation Partnership Project) agreed earlier this year to release 5G NSA (Non-Stand Alone) specifications in March 2018. NSA incorporates the advanced radio technologies of 5G but utilizes 4G LTE infrastructure, a step that is intended to allow commercial services to launch as early as the end of 2018. Full 5G specifications (called 5G SA or Stand Alone) would follow in late 2018 and include the core network improvements needed to fully support IoT devices, enhanced routing protocols, and other advances.

Once the full set of standards is complete, there is still a lot of work that needs to be completed before 5G becomes a reality. Chip designs will need to be finalized, handset manufacturers will need to produce devices, and service providers will need to deploy base stations, most likely beginning in major city centers. The “2020” included in the title of the ITU’s report named above is not only part of the document number, but also the calendar year when true 5G is likely to become commercially available.

Wes Simpson is the president of Telecom Product Consulting. He can be reached via TV Technology.

Figuring out the amount of bandwidth a video signal requires on an IP network isn’t terribly hard, but it does require some familiarity with the underlying technologies and packet formats. Getting the correct answer is important for tasks such as estimating the number of videos that can be carried over a given network connection or for calculating the costs of a long-haul connection to carry signals between two facilities.

In this column, we’ll look at how much bandwidth will be consumed for a 1080p59.94 video signal when it is transported over two popular formats for uncompressed IP video transport that are available today.

The first format is SMPTE ST 2022-6, which was originally designed for moving uncompressed signals over a long-haul network, including all of the embedded audio and any other signals contained in the HANC and VANC spaces. This format is still popular today because it is very easy to take an SDI signal source (SD, HD or 3G), convert it into ST 2022-6 for transport over an IP connection, and get back exactly the same SDI signal at the destination without changing a single bit. It has been widely implemented by a number of equipment suppliers, and interoperability has been proven at several industry events, including VidTrans, which is hosted by the VSF (Video Services Forum).

The other format is newer, but it allows transportation of each type of media essence (video, audio, etc.) as a separate IP packet stream. This approach eliminates the need to embed and de-embed audio and other signals into SDI streams for transport and, as shown in the following calculations, reduces the amount of IP network bandwidth needed for video transport. The VSF TR-03 recommendation is based on RFC 4175, which takes groups of pixels and directly maps them into RTP packets. This recommendation is expected to evolve soon into SMPTE ST 2110-20, which will use a similar packet format.

PACKET PAYLOADS

The first step in calculating signal bandwidth is to figure out how much media essence can be transported in each packet. For ST 2022-6, this step is easy—each packet carries a fixed payload of 1376 bytes. For VSF TR-03, several alternatives are possible, so to simplify calculations, each video line consisting of 1,920 pixels will be divided into four equal parts of 480 pixels each. With 4:2:2 sampling, each pair of pixels requires four 10-bit samples (two luma and two chroma), which equates to 40 bits or 5 bytes. Thus, 480 pixels will occupy (480/2)*5=1200 bytes.

PACKET HEADERS AND OVERHEAD

As shown in the calculations in Fig. 1, at each layer as packets move through the protocol stack, new headers are added. For ST 2022-6, a High Bitrate Media Header is added. For TR-03 (and soon ST 2110-20) an 8-byte payload header is used. Then, the 12-byte RTP header and 8-byte UDP header are applied, followed by an IPv4 header of 20 bytes. At the Ethernet layer, the standard Ethernet header of 14 bytes is often extended by a 4-byte VLAN label, and the required 4-byte Frame Check Sequence is appended to the packet, for a total of 22 bytes of overhead. When transmitted over a standard path, each Ethernet frame is preceded by an 8-byte preamble and followed by an inter-frame gap equal in duration to 12 bytes, for a total overhead equal to 20 bytes in duration.

PACKET RATE CALCULATIONS

Once the size of each packet is known, the other factor needed to calculate a signal’s bandwidth is the number of packets per second. This needs to be calculated using the original signal rates. 

For ST 2022-6, since the entire 1080p59.94 payload is transported, this calculation must be based on the full video frame. With 2,200 samples per line, 1,125 lines per frame, and 20 bits per sample (in 4:2:2 10-bit sampling), the total number of bytes per frame is 6,187,500. With 1376 bytes per packet, this translates to 4,497 packets per video frame. At 59.94 frames per second, the total packet rate is 269,550 packets per second.

For TR-03/ST 2110-20, only the active video area is transported. Since each packet carries one-quarter of a video line, one full video frame will require 4x1080 = 4320 packets. At 59.94 frames per second, the stream will consume 258,941 packets per second.

To get the total bit rate, all that remains is to multiply the packet rate by the size of the packet in bits. As the bottom row of Fig. 1 shows, the bandwidth of a 2022-6 signal is about 200 Mbps higher than the nominal 2.97 Gbps required by a 1080p video, whereas the TR-03/ST 2110-20 is almost 300 Mbps less than the raw SDI.

WHAT ABOUT AUDIO?

The only other high-bandwidth signals that are commonly found in a modern production facility are audio signals. In ST 2022-6, the audio signals are carried inside the SDI payload, so there is no extra bandwidth required for audio (provided the number of audio channels is less than what the SDI can carry).

In TR-03/SMPTE ST 2110, more bandwidth will need to be allocated for audio, although, with a 48 KHz, 24-bit stereo signal occupying less than 3 Mbps, audio streams are generally not a major burden on a gigabit-class network.

A NOTE OF CAUTION

The actual amount of bandwidth allocated (i.e. the CIR or Committed Information Rate) in any network connection that carries an IP video signal needs to be greater than the raw bit rate calculated in this article. In particular, due in part to the bursty nature of video (blocks of pixels with gaps where the VANC would be), additional bandwidth should be provisioned through each network hop above and beyond the amounts calculated in this article. Since the recommended amount of added bandwidth is currently being studied by the SMPTE committee, this will have to be the subject of a future column.

Wes Simpson is active in standards development and technology training. Please visittelecompro.tvfor more information.