July 27, 2017

Can You Hear Me Now?: A History of the Development of the Intercom (Part 3)

Comedian Bob Newhart's first record album, "The Button-down Mind of Bob Newhart", has a hilarious routine called "The Khrushchev Landing Rehearsal". Newhart's routine centers on a television director in the TV control room, trying to direct the run-through of Khrushchev's airplane landing at Idlewild Airport. Soviet Premier Nikita Khrushchev visited the United States in September of 1959. One of Newhart's gag lines makes fun of then-President Eisenhower's love of playing golf. As Khrushchev approaches Eisenhower to shake hands, Newhart says over the intercom in frustration, "Somebody take the putter from Ike!"

Of course, there was no run-through of this event. News events aren't rehearsed. However, Newhart does a great job of portraying what you would hear on a typical partyline intercom in a broadcast environment. His routine also shows how important to the entertainment industry a functional intercom system is. Television, live shows and motion picture production could not be done without intercom systems. 

After telephones had become common, telephone-type intercom systems started to be used in live theater. There were standard handset-based phone instruments located backstage, in offices, out front in the prompt booth, as well as other areas. But there was still a need for stage managers to be able to talk to their technical crews, especially the spotlight operators. Many theater companies had technical people who built specialized voice communication systems on their own. A one-way talk system was all that most live venues had to use. 

Crew members wore headphones to hear the manager. They could not hold a handset and still perform their duties so they weren't able to talk to the stage manager, or each other. Most of the early headset products used by telephone operators and other users were pretty clunky. Many of them consisted of a pair of headphones and a large, round, black curved tube that sat on the users chest. At the bottom of this tube was a microphone. It was secured by a leather strap that went around the user's neck. 

As time passed, telephone equipment became more sophisticated and cheaper to purchase and creating a partyline system became easier to do. Headsets had become smaller and lighter, most likely due to the demand on part of the telephone company operators. Wearing one of the early headphone/speaking tube microphones for eight hours at a stretch was exhausting and uncomfortable. When television broadcasting began in the late 1940's, the need for an intercom system to connect directors, floor people, camera operators, video switchers, and sound board operators became a necessity instead of a luxury. 

Some TV pioneers tell stories of the pre-intercom days when crew members tried to communicate exclusively with hand signals, much like baseball players do. For a baseball team, this method works well, but in a production environment, it's much too slow and doesn't allow for sudden changes that take place.

Like early theater, most television station's engineering departments built the intercoms used within their facilities themselves. These were typically based on existing telephone technology - by using a power supply that would source the considerable current, multiple users could connect to the same line without lowering the voice level too far to be heard clearly. And the early partyline system was born.

By this time, headsets had evolved into smaller and lighter versions. Wearing a headset now didn't weigh down the user and left their hands free to run a camera, video switcher, or perform whatever duties that person needed to do.

When entertainment production became increasingly complex, intercoms had to also be more complex in their functionality. Large productions needed more than one communication circuit and users frequently needed the ability to switch themselves from one circuit to another. Many venues and stations used one intercom system that covered the entire facility. Local user panels were installed in areas like editing rooms, offices of programmers, and even places like the General Manager's office or the commercial traffic office. For systems of this size, matrix-type switching and local station re-routing were necessary to satisfy all of the requirements. Having the staff engineering people construct and deploy the intercom system just wasn't an option. The operational needs of the system became too complex. 

This was, and still is, true in large-market TV stations that cover many floors. For example, I once worked in a station that had the newsroom on one floor, the studios and engineering department on another, and the sales, traffic and accounting departments on a third floor. One intercom system connected the entire facility and many of the rooms and offices on each floor had an intercom panel. All of the studios were equipped with multiple beltpacks because there were always numerous live productions underway.

Today, sophisticated intercom systems have become so common that much of the modern technology available is taken for granted. The ability to reconfigure systems on the fly, having instant reliable access, high-quality audio, and features such as interruptable foldback (IFB) for cueing and orders from the director, make the modern intercom systems the backbone of any production. Without this capability, producing everything from sporting events to live plays to opera to television shows would not be possible. 

In case you missed Part 1 and 2 of this series:
    Part 1: Totally Tubular: A History of the Development of the Intercom
    Part 2: Mr. Watson, Come Here: A History of the Development of the Intercom

Paul Black is a freelance writer and broadcast engineer in Northern California. He holds a Certified Professional Broadcast Engineer certification from the Society of Broadcast Engineers and an FCC Lifetime General Class Operator License. He is a licensed amateur radio operator (call sign N6BBZ) and has worked for several broadcast companies, including Bonneville Broadcasting, RKO General Broadcasting, and CBS Television. Visit his website at www.paulblackcopy.com

July 5, 2017

Mr. Watson, Come Here: A History of the Development of the Intercom (Part 2)

In the first part of this series, we discussed how the words "Internal Communication" led to the now common term, Intercom. We also talked about how the first intercom systems were known as voice tubes; tubes that carried voices acoustically. Although these were very crude systems compared to later electrical products, they did actually work. Some types of voice tubes are still in use today.

Like many other technologies, electrification changed intercom products radically. Before the electrical intercom could exist, one invention had to proceed it.

The telephone.

Bell making the first telephone call

When Alexander Graham Bell sat in one room in his Boston home, raised a transmitter tube to his mouth and then said, "Mr. Watson, come here. I want to see you", he was talking from that room down to his assistant in the basement, Thomas Watson. Watson, an engineer and employee of Bell's, once claimed that Bell had spilled battery acid on his leg and was genuinely calling for help. However, Watson later said that he could not be certain if this took place during the first test or in subsequent research. In any case, Bell documented the event in his journal on March 10th, 1876, making the first words spoken over what would later become the telephone, revolutionizing communications around the world.

One of the first systems ever patented was built by the Kellogg Switchboard and Supply Company in 1894. It's main purpose was for use in apartment buildings. Some early apartment buildings had a pull-wire, then later electrical, signaling system. A prospective visitor would pull a handle, or push a button on a panel, which would then signal the tenant that someone wanted inside. However, because they couldn't see or talk to who was signaling them, they ran the risk of letting in someone they didn't know, like a burglar.

As is true today, the more upscale apartment buildings in major cities had doormen to guard the building. Prior to the intercoms being available, these men would have to lock out the person asking for entry and go up to the tenant's apartment to ask if the tenant wanted to see the visitor or not. As apartment buildings became larger, this process became frustrating to both the tenants and the visitors.

Early Kellogg Phone

The Kellogg system added a telephone-type circuit to the signal panels. An earpiece and mouthpiece resembling the old-fashioned candlestick telephones allowed the tenant to talk to the visitor directly, instead of having the doorman run up and down to ask the tenant in person.

In the late 1930's, an engineer named Allan C. Bernstein started a company called Adams Laboratories that took the apartment intercom idea into the workplace. Ultimately, Adams Laboratories became the manufacturing arm of Executone Corporation, with Bernstein as President. The first product was a simple two-station system, called the "boss-to-secretary" system. This migrated into further point-to-point wired systems, including one of the first "patient-to-nurse" hospital communication systems ever used. Over his lifetime, Bernstein developed and held many patents until his death in 1987.

RMS Queen Mary's Loudaphone

One unique intercom system for use in noisy environments was made in England in the 1930's. Called the Loudaphone, it was installed in places with high background noise, such as trains or ship engine rooms. For example, the RMS Queen Mary had a Loudaphone system in use in the engine room areas.

After the Bell System monopoly was established in the early 1920's, the local telephone companies owned by Bell, then later American Telephone and Telegraph (AT&T), were the only convenient source of equipment and services for regular land-line telephone systems. This gave them a perfect opportunity to also be the provider of intercom systems.

Multiple desk phone intercoms

In old movies from the early part of the twentieth century, some of the scenes set in offices show a desk with two or three phones. Many times, one of the phone would be a part of an intercom system within the building. It wasn't until more sophisticated systems came along that the phones used for outside lines also had an intercom tied to them. However, not all businesses took advantage of this capability. Today, this is still the case.

As time passed, the needs of the marketplace caused manufacturers to make a lot of progress on new features. Speaker-based intercoms, such as those used in naval ships in WWII, became common in the commercial world. Since these operated like a speakerphone, they didn't require a handset. More than one person at a time could participate in the conversation. Noise-cancelling circuits for loud areas, public address capabilities, and other similar modern features made it easier for end users to find a system that suited their needs.

Wires, wires and more wires

One large cost factor in intercom installation was the wiring that had to be run inside the building. Pulling wire is extremely labor-intensive and it can be the most expensive part of an installation. To help with this, manufacturers came up with wireless systems that function like speakerphones, but use radio frequencies instead of wiring. Another unique solution that was developed became know as a Powerline system. A Powerline is when the intercom hardware uses the wiring that provides power to the outlets as a path for the signal.

In our next installment, we'll talk about the current state-of-the-art intercom technology and address two of the largest users of intercom products - the entertainment and broadcasting industries.

In case you missed Part 1 of this series:
    Part 1: Totally Tubular: A History of the Development of the Intercom

Paul Black is a freelance writer and broadcast engineer in Northern California. He holds a Certified Professional Broadcast Engineer certification from the Society of Broadcast Engineers and an FCC Lifetime General Class Operator License. He is a licensed amateur radio operator (call sign N6BBZ) and has worked for several broadcast companies, including Bonneville Broadcasting, RKO General Broadcasting, and CBS Television. Visit his website at www.paulblackcopy.com

June 27, 2017

Totally Tubular: A History of the Development of Intercom (Part 1)

People have always needed to communicate over long distances. From the smoke signal used by Native American tribes, to the telephone, to email via the Internet, the ability to communicate at a distance is a need that goes back through the millennia.

Equally so, communicating over smaller distances has also been important. Talking from one floor to another in a single building, or within any large area (such as inside a warehouse) requires some form of internal communication. 

From "internal communication", we get the now common term of "Intercom". This term is generally used as the name of a device or system that allows people to speak to each other from one room to another, or across a large open area inside a structure.

Before electrical systems were common, the usual way to get someone's attention was by some form of signaling. As an example, communicating in a large mansion-style home from the parlor to the servant's quarters was done by pull ropes that were connected to bells that would ring - when the rope was pulled, the bells would ring, grabbing the attention of the servant. When electrical signaling devices became more commonplace, these pull ropes were replaced with wired pushbuttons that would trigger a buzzer that would sound in the servant's quarters. Whether with the pull ropes or the wired pushbuttons, the servants were only alerted that their services were needed; therefore they needed to walk from one area of the house to the other to find out what was needed.

Being able to actually talk to a person in another part of a building would be more efficient. The result of this need of efficiency was the development of an acoustical communication system, or more simply, a voice tube. A voice tube is a hollow tube or pipe that was run from one place to another, allowing people to talk back and forth to each other from different ends of a building.

The idea for the original voice tube began back in the early nineteenth century by the French scientist, Jean-Baptiste Biot. His unusual choice of laboratory in which to test his theory - the water pipes of his home city of Paris. From his experiments, Biot discovered that smaller pipes carried sounds over amazingly long distances. Larger pipes just didn't work as well for carrying sound.

In 1849, Scientific American magazine published an article describing what they called as an acoustic telegraph - a tube made of gutta percha (a latex-like material that's produced from the sap of trees that grow in Malaysia). The article claimed that a tube made of this material and of the proper size could send voices for several miles.

Later voice tube developments included work done by Antonio Meucci, an Italian immigrant scientist who built an acoustic speaking tube system in his home in New York. He later was attributed for his work on what eventually became known as the telephone.

Early aircraft models were usually of the open-cockpit design and were extremely noisy. Instructors had to scream at their flying students in order to be heard above the sound of the engine and the slipstream noise. So, in 1917, at a flying school in Gosport, England, an instructor named Robert Raymond Smith-Barry sought to overcome this problem with some rubber tubes that had a funnel on one end, and a pair of primitive headphones made of cloth and rubber on the other end. The student wore the headphones and the instructor spoke into the funnel. Smith-Barry called his invention the Gosport Tube, after the town the school was in. This became a commonly used item for flight instruction until as late as the 1930's when closed cockpits and electrical intercoms became the standard in aircraft.

In 1926, the Bureau of Standards of the United States Department of Commerce issued a paper that could be purchased by the US Government Printing Office for a pricey 15 cents, entitled "Transmission of Sound Through Voice Tubes". It is an exhaustive study of the physics of sending sound via hollow tubes, including photographs and mechanical drawings of the tubes used in the test procedures. The US Navy actually initiated the request to the Bureau of Standards for these tests to be done. The Navy is one of the prime users of acoustic voice tube technology, and that is true even to this day. Currently, modern Navy landing craft (or LCU's) use voice tubes to communicate from their upper deck to control centers below deck. Other naval vessels also use them, as well as merchant marine cargo ships.

The major advantage of a voice tube system over an electrical communication system is its simplicity and reliability. Voice tubes are impervious to the problems plaguing electronic systems, such as power failures, broken or shorted wiring, and invasion by moisture. So, despite the sophistication of modern electrical intercoms, acoustic speaking or voice tubes remain in use. Because of this, voice tubes make these a good choice for these applications.

In our next installment, we'll look at the rise of the electrical intercom systems, how they developed from the days of the invention of the telephone, and why some of the first applications of these products were for safety and security purposes.

Paul Black is a freelance writer and broadcast engineer in Northern California. He holds a Certified Professional Broadcast Engineer certification from the Society of Broadcast Engineers and an FCC Lifetime General Class Operator License. He is a licensed amateur radio operator (call sign N6BBZ) and has worked for several broadcast companies, including Bonneville Broadcasting, RKO General Broadcasting, and CBS Television. Visit his website at www.paulblackcopy.com

June 20, 2017

Do You Need a LAN or a WAN?

When designing an IT network topology, you might want to consider a Campus Area Network (CAN), a Metropolitan Area Network (MAN) or a Tiny Area Network (TAN), but most certainly a Local Area Network (LAN) or Wide Area Network (WAN) will do the job. All of these networking schemes utilize the same switches and file transfer technologies and are used to connect computers and devices, allowing them to communicate in a specific geographical area or region.

You might be most familiar with the computer networking terms like LAN or WAN, which are thrown around a lot in conversations about setting up IT networks and collaborative environments, but what do they really mean to your organization? 


A Local Area Network (LAN) is a group of computers and network devices that are connected together, usually within the same building. Examples could be a small office or production facility, a single building or multiple buildings located on campus. 


A Wide Area Network (WAN), as its name implies, connects several LANs together (whether nearby or in different parts of the world) and is typically used by an enterprise-level installation (a corporation or organization) or local governments that make civic information easily accessible to the public.


The technology employed - routers, servers, cabling, desktop clients for users - is high speed and relatively expensive. LANs tend to use high-speed connectivity technologies like Ethernet (CAT5/6 cabling) or Token Ring (all computers are connected in a ring or star topology). WANs most often use technologies like MPLS, ATM, Frame Relay and X.25 for connectivity over long distances.

LANs use Layer 1 devices, like hubs and repeaters, and Layer 2 devices, like network switches and bridges. WANs use Layer 3 devices, such as routers, multi-layer switches and specific devices like ATM or Frame Relay switches.

One LAN can be connected to other LANs over any distance via telephone lines and radio waves, while computers or other networked devices connected to a WAN are often connected through public networks, such as the telephone system. They can also be connected through leased lines or satellites.

The major advantage of a LAN is the speed it can reach. With a LAN, it isn't uncommon to see technology that supports 1Gbps file transfers. Most agree that a LAN can operate up to 30x faster than a WAN. The further the distance, the slower the network. However, the major disadvantage with a LAN is that it is only good as far as you can reach an Ethernet cable or WiFi signal. Simply put, you cannot buy an Ethernet cable that will reach throughout an entire building, and a WiFi connection rapidly deteriorates as you get further than a few dozen miles away.

A WAN connection is generally harder to setup, but there are many creative ways to do so. One very common way is renting a line from an Internet service provider and using their network (that's already connected to the entire world) and connecting Point A to Point B. Another way to do a WAN is connecting the devices with various wireless technologies, like cellphone towers or satellites. As you can imagine, all of these are much harder to create than setting up a LAN, and almost always demands high level professional setup and maintenance.


LANs are generally more secure than WANs, but, of course, WANs enable more widespread connectivity. And, while LANs tend to be owned, controlled and managed in-house by the organization where they are deployed, WANs typically require two or more of their constituent LANs to be connected over the public Internet or via a private connection established by a third-party telecommunications provider. As for actual reliability, LANs tend to have fewer problems because there are less systems to deal with. A WAN tends to be less fault tolerant as they consist of a large number of disparately located systems that have to be reliably connected.


One of the big disadvantages to implementing a WAN is the cost. Having a private WAN can be expensive because of the technology required to connect two remote places together. However, WANs using public networks can be setup very cost-effectively using Virtual Private Network (VPN) hardware and software, which allows a desktop computer to transparently connect to a remote network as if you were physically attached to that network. For security, the communication link between your computer and the remote VPN hardware is encrypted while using the VPN.

The maintenance costs are often lower with a LAN because it covers a relatively small geographical area, while maintaining a WAN is difficult because of its wider geographical area.

In the end, it's clear that those implementing a network - whether a LAN or a WAN - should take a hard look at how they plan to use that network and who will use it. Each topology has its advantages and disadvantages that can affect an organization's productivity significantly. Which one you choose ultimately depends on costs and your business model.

June 1, 2017

The Origin of TCP/IP

These days virtually everyone is talking about Internet Protocol (IP), the standard encoding scheme for sending audio and video files (as IP packets), either over the Internet directly or over an Ethernet-based CAT5/6 cabling network. The cable itself is favored because it is much smaller and lighter and therefore much easier to work with when building new A/V facilities. But where did the IP scheme as we know it today come from?


First, we have to look at Transmission Control Protocol/Internet Protocol (TCP/IP), which basically describes a protocol that will work on any sort of computer and operating system for transportation of packets of data across the internet or studio network between different systems.

Early concepts of packet networking were developed at several different research labs in the US and overseas, initially for military use. In the late 1970's, a set of networking protocols that allowed two or more computers to communicate, known as TCP/IP, were developed by The Defense Data Network, part of the Department of Defense, for widespread industry use across its Advanced Research Projects Agency Network (ARPANet). ARPANet was an early packet switching network and the first network to implement the protocol suite known as TCP/IP. Soon, several other TCP/IP prototypes were developed at multiple research centers between 1978 and 1983. However, the migration in the US of the ARPANet to TCP/IP was officially completed on January 1, 1983, when the new protocols were permanently activated across what has since become the World Wide Web.

Source: wikipedia.org

For production professionals, TCP/IP has become very important to audio file delivery and networking because it allows the flexibility to route resources to any part of a facility (or remote location) from a centralized position. It also facilitates the development of Audio-over-IP (AoIP) networks that allow convenient control and monitoring of equipment and systems, and the rapid transfer of audio and firmware files between components.


With the use of TCP/IP, the amount of information a single cable can carry has increased from a few thousand bits per second in the 1960's to a few billion bits per second today. Regular affordable connections in every day information systems now carry one or more gigabits of information in a single fiber cable over distances spanning many miles. This bandwidth is enough to transport hundreds of high quality audio channels, replacing hundreds of pounds of cabling in conventional systems. More importantly, the functional connections in a networked audio system can be designed separately from the physical connections in the network, due to the flexibility of the IP scheme.

The functionality opens up a wide array of exciting possibilities for the audio industry: any number of I/O locations can connect to the network anywhere in the system without the limitations of bulky cables, leaving the actual connections to be managed with easy-to-use software. Control signals can be included in the network without additional cabling. Computers can use the network to control and monitor audio devices, such as digital mixers and DSP engines. Video connections can also be included using affordable IP-controlled cameras. 

IP is allowing users to send high-quality audio feeds over long distances. This is also known as Audio Contribution over IP (ACIP), which enables programming contributions from outside members of the team as if they were in the next room.


Clearly, the Internet and IP have changed the way audio production is performed across a wide variety of applications. Time and space limitations are no longer valid. Professional productions now benefit from the ability to bring in all types of sources from a variety of disparate locations to support and control an array of devices.

The inevitable merging of computer networking technology and audio distribution has arrived. It's time to re-examine the assumptions and concerns (eg, latency, security, etc.) that are holding some professionals back from choosing Audio-over-IP solutions and just get on with it. IP packets can be easily sent and received in a variety of ways that help streamline workflows of all types. It also potentially enables more productivity with less people.

Early AoIP networks were plagued with dropouts, pops and clicks, with most devices limited to a paltry 10 Mbps bandwidth. Some thought that IP packets were never going to be fast enough to deliver real-time audio. With modern advances in networking connectivity, now we know you can.

Network speed is no longer a practical limiting factor. Gigabit speeds and modern switching technology ensure virtually zero packet loss under real life conditions in copper or fiber optic networks, while providing ample bandwidth for hundreds of channels of audio and other data at each node. Switched networks isolate traffic at each port, permitting thousands of channels to exist without conflict on a single network without dropouts or errors. 


The effect that the Internet Protocol has had on today's audio professional cannot be overstated. TCP/IP ensures that data will get to the correct destination and received in the correct order. We're living in a time of massive online growth. IP has helped make sense of it all and, really, saved the day by making our lives a whole lot easier. And the best part: an IP network will be relevant and easily upgradable for years to come.

May 19, 2017

Communication Types: Simplex vs Duplex

There are many different ways of communicating, but what exactly do the terms Simplex and Duplex mean? Here's a brief rundown of what these terms mean.

Simplex Transmission

In Simplex Transmission, data flows only in one direction - from the sending device to the receiving device. Simplex Transmission is used only when the sending device does not require a response from the receiving device. 

Example: A microphone to a loudspeaker

Half-Duplex Communication

Half-duplex communication allows two-way conversations, one-way at a time, such that one person cannot interrupt the other. 

Example: A walkie-talkie

Full-Duplex Communication

Full-duplex describes bi-directional communications all the time. Regular communications between individuals conversing face to face is full-duplex. In other words, you can talk and listen simultaneously. Full-duplex communication allows simultaneous two-way conversations where one person can interrupt the other.

Example: Two people having a conversation.

May 17, 2017

Audio-over-IP Networking Takes Many Forms

While everyone agrees that networking audio products to create flexible workflow infrastructures is indeed the best way to distribute and leverage the multiple signals used for a video production or broadcast facility going forward, there's a bit of a debate about the most efficient way to do it. We see and hear a lot of acronyms and fancy names thrown around when it comes to networking, that is, sending audio signals over an Internet Protocol network (AoIP), but what's it all mean?

The two major competing formats - Dante (Digital Audio Network Through Ethernet, a proprietary but widely accepted protocol from Australia-based Audinate) and AES67 (a more open industry initiative) - similarly convert audio signals to IP and enable complete flexibility in where audio equipment can be located and how signals travel around a facility or live production. Yet, they also have their differences. Choosing the right one has led to a bit of confusion among customers seeking to set up networks that allow individual pieces of audio equipment (microphones, mixing consoles, routers, audio processors, etc.) to communicate with each other in new and innovative ways that have not been possible before.

Both the AES67 and Dante AoIP protocols convert audio signals to IP 
and enable complete flexibility in where audio equipment can be
located and how signals travel around a facility or live production.

Many applications and proprietary networking benefit from both protocols and, interestingly, both are supported by each other in many systems cases - such as audio mixing consoles from companies like Solid State Logic and Wheatstone. And the benefits for users are potentially huge, allowing broadcasters, production studios, houses of worship and other venues to combine studios and resources as a result of IP audio networking. The flexibility of these systems makes it possible for them to change the studio from one network affiliate to another in a matter of minutes. That includes reconfiguring everything, from remote venue to mix-minus and talent mics. No longer do users have to keep dedicated studios for each show or affiliate. One, very flexible studio can serve multiple purposes.

On a practical level, IP audio is much easier to deal with and far more scalable for doing the fast-paced productions that go on in production environments these days. Mic sharing, studio swapping, and setting up IFBs are straightforward and quick, which means a good return on investment because you're able to streamline production and reduce hardware costs in many cases. 


Dante consists of software, hardware, and network protocols that deliver uncompressed, multi-channel, low-latency digital audio over a standard Ethernet network using Layer 3 IP packets. It consists of a suite of technologies: an easy to integrate Audio-over-IP networking implementation, along with well-defined developer tools and software applications. This provides manufacturers with a complete toolkit from which to build networked products, plus it gives end users everything they need to easily design, use and manage systems with audio networking.This is very different from only having a published standard, which still requires additional building blocks, which may result in very inconsistent implementations. 

Layer 3 IP Networking Packet

Dante has been licensed by approximately 350 companies, which have in turn produced more than 1,000 products incorporating the AoIP technology. Audinate produces Dante modules, chipsets and reference designs to fit virtually any type of audio product, from tiny, cost-effective single-channel devices to massive consoles with hundreds of channels. And they all work the same way, with the same tools. Dante offers, by far, the widest uptake and the most defined advanced feature-set of the current technology options. It represents a reliable, high-capacity, ultra-low latency audio network with plug-and-play discoverability, full redundancy, advanced API, and excellent value for the customer, according to the company.

Dante is also designed to work with standard network switches, allowing it to be easily integrated with existing IP infrastructure in a facility without the need for special network hardware. Interestingly, Dante also incorporates support for AES67, allowing users to connect to other audio networks without disturbing the routing or performance of the Dante network in any way. Finally, Dante Controller software provides users with audio-specific network management tools that allow them to monitor clock health and latency in real-time. 


Many new products are now being introduced that natively support AES67. It is an AoIP interoperability standard, developed by the Audio Engineering Society and published in September 2013, which is intended to provide the audio transport and clocking requirements for different AoIP technologies to interchange audio. It does not strictly define how to control and manage these connections. The details of the configuring of these connections are then determined by each underlying technology, manufacturer implementation or a control system deployed that manages multiple technologies.

Demonstrations of the AES67 networking format have been held at several audio industry
conventions to show how the open standard helps manufacturers, integrators and end users alike.

The AES67 standard is what's called a "Layer 3" protocol software suite based on existing standards that is designed to allow interoperability between various IP-based audio networking topologies, like Ravenna and yes, Dante. It also identifies commonalities with Audio Video Bridge (AVB) and specifies real world AVB interoperability scenarios. AES67 is designed to ensure interoperability between previously competing AoIP systems and long-term network interoperation between systems. Since its publication, AES67 has been implemented independently by dozens of manufacturers and adopted by many broadcast and production facilities.


Ravenna is yet another audio networking technology for real-time transport of audio and other media data in IP-based networked environments. It was first introduced in September of 2010 at the IBC convention in Amsterdam and is compatible with AES67, since all relevant mechanisms, protocols and formats used for synchronization, transport and payload mandated by AES67 are fully supported. Ravenna is based on protocol levels at or above the Layer 3 protocol. All protocols and mechanisms used within Ravenna are based on widely deployed and established standards. Ravenna can operate on most existing network infrastructures using standard networking technology. Performance and capacity scale with network performance. Ravenna is designed to meet requirements for low latency, full signal transparency and high reliability.


So, does the audio industry have to standardize on one format? Not necessarily, according to those with experience in audio networking.

"There is space for different technologies and standards within the audio industry and it is important to recognize the difference between the two," said Tom Knowles, Product Manager of Broadcast Systems at Solid State Logic. The company's System T broadcast audio production environment is based around Audinate's Dante AoIP technology and also incorporates the AES67 standard.

"Different AoIP technologies have their own benefits and compromises. The value of a standard, such as AES67, is interoperability between the different AoIP technologies," said Knowles. "Those technologies still exist to provide added feature sets, while standards provide a level of interoperability for the user when a single technology is not used."

Today's standard industrial IT infrastructure has already overtaken the technology of AES/EBU, MADI and TDM routers in terms of performance, cost and flexibility. Networked systems deliver huge improvements in infrastructure flexibility, better scalability, and significantly lower costs. Broadcasters and systems integrators can expect more choices in selecting interoperable equipment and solutions from a range of suppliers and will have more scope to manage and control these systems.

In the end, improved flexibility, interoperability, and features like discoverability will mean the broadcasters can adapt their systems as their own needs evolve and demands change. Freedom from expensive central routing hardware means that budgetary and inflexible infrastructure obstructions to improvements, change, and renewal are removed. Networked control allows multiple operators to share core processing power, opening up many possibilities for collaborative and multi-role production.

January 26, 2017

Exit, Stage Left

If you've ever worked on a stage, read or wrote a script, or have ever watched the Hanna-Barbera cartoon featuring the wanna-be stage actor, Snagglepuss, you've most likely have heard the phrase, "Exit, Stage Left". But what exactly does that term mean?

Here's a quick glossary of stage terms that describe the locations and directions in a theater. Many of these terms actually come from the descriptions of stage actions in a play script.

HOUSE LEFT: the left side of the auditorium from the spectator's viewpoint facing the stage; toward or at the left side of the auditorium from the spectator's viewpoint facing the stage. Also known as Front House Left or Rear House Left.

HOUSE RIGHT: the right side of the auditorium from the spectator's viewpoint facing the stage; toward or at the right side of the auditorium from the spectator's viewpoint facing the stage. Also known as Front House Right or Rear House Right.

STAGE LEFT: the left side of the stage from the actor's viewpoint facing the audience; toward or at the left side of the stage from the actor's viewpoint facing the audience.

STAGE RIGHT: the left side of the stage from the actor's viewpoint facing the audience; toward or at the left side of the stage from the actor's viewpoint facing the audience.

DOWNSTAGE: the part of the stage nearest the audience; toward or at the front part of the stage, nearest the audience. Often used in conjunction with Left or Right: Downstage Left, Downstage Right.

UPSTAGE: the part of the stage farthest from the audience; toward or at the rear part of the stage, farthest from the audience. Often used in conjunction with Left or Right: Upstage Left, Upstage Right.

OFFSTAGE: away from the stage center; toward or at the part of the stage that is out of view of the audience. Often used in conjunction with Left or Right: Offstage Left, Offstage Right.

ONSTAGE: toward the stage center; toward or in the part of the stage that is in view of the audience.

OUT: away from the stage floor and toward the gridiron, referring to vertical movement of scenery, etc.

IN: toward the stage floor, referring to vertical movement of scenery, etc.

Source: Theatre Projects Consultants