Industry Insights: Modernizing intercom and audio mixing workflows

In this installment of our Industry Insight series on control room technology, industry experts share how broadcasters are adapting intercom systems and audio mixing workflows for IP, cloud and hybrid environments.

From ST 2110-based comms networks to software-based mixing consoles, the conversation highlights the convergence of audio routing, intercom and mixing into unified, scalable systems. Vendors explore the growing demand for flexible, decentralized setups that support everything from distributed remote production to immersive audio monitoring – all while maintaining reliability and clarity under live conditions.

As broadcasters reimagine their control rooms, intercom and audio systems are no longer isolated components; they’re integral parts of a fully networked production ecosystem.

Industry Insights: Production control room strategy in the era of software-defined broadcast

Key takeaways from this Industry Insights roundtable

• Audio is converging: Broadcasters are unifying intercom, audio mixing, and routing through IP standards like AES67 and ST 2110 to streamline operations.
• Software is taking over: Software-based mixing solutions offer more flexibility and scalability than hardware consoles, especially in hybrid environments.
• Remote-ready workflows: IP standards now support WAN-compatible audio control, enabling permanent remote production setups across locations.
• Mixing demands versatility: Modern consoles must support simultaneous output formats, from stereo to immersive, while remaining intuitive and accessible.
• Comms require integration: Broadcasters expect intercom systems to interoperate directly with mixers and routers, reducing system fragmentation.

What are broadcasters prioritizing when upgrading intercom systems?

Adam Salkin, senior solutions architect, Diversified: Broadcasters are prioritizing connectivity to 2110 audio flows and compatibility with audio mixers. Intercom audio should no longer need to be a separate system from an audio mixer.

Marc Wilson, managing director, Glensound: An intercom system must now include the option of Dante or Ravenna AES67 network audio, to allow interfacing across the network.  As channels can be freely assigned, screens per channel are increasingly required to provide source/destination information taken directly from the Dante Controller.

Martin Dyster, VP, business development, Telos Alliance: The landscape is mixed depending on the broadcaster, and the type and complexity of content they are producing. Where we sit in the market with Telos Infinity, our customers are usually looking for flexibility, value for money, and the ability to scale as their business grows. When we replace or enhance legacy systems, it is often because the newest version of that older technology doesn’t provide these benefits.

Dave MacKinnon, VP, product management, Clear-Com: Broadcasters are focusing on flexibility and scalability to support hybrid workflows that blend in-studio and remote production environments. Reliability and ultra-low latency are non-negotiable for live broadcasts, and intuitive user interfaces help reduce operator training and errors during fast-paced productions. Our newly debuted, award-wining FreeSpeak Icon beltpack exemplifies these priorities by offering expanded capacity, extended range, and exceptional audio clarity, meeting the demands of dynamic broadcast teams.

Joyce Bente, president and CEO, the Americas, Riedel Communications: Broadcasters upgrading intercom systems are increasingly prioritizing high-quality audio, rock-solid reliability with redundancy, remote and cloud-enabled workflows, scalability, and standards-based interoperability. They demand clear, low-latency sound with advanced noise suppression to ensure communication is intelligible even in noisy live settings. Reliability is achieved through redundant network designs like meshed or ring topologies and hardware failovers to prevent downtime, while support for remote productions via IP and cloud infrastructure — using NMOS for integration with broadcast control systems and secure mobile app access — has become essential.

Advertisement

How have IP workflows influenced intercom reliability, latency, and voice clarity during live productions?

Martin Dyster, VP, business development, Telos Alliance: IP is a means of connectivity for most intercom systems now, but for Telos Infinity, it is at the heart of our natively IP system. Reliability is paramount, and a properly designed media LAN gives you the assurance that your comms will never let you down. AES67/ST2110-30 is 48K and 24-bit, so communications audio and contribution are the same, meaning that voice clarity is indistinguishable between subsystems. Finally, latency on an IP LAN is a non-issue with 125 µs and 1 ms packet types the norm, provided proper synchronization via PTP is employed.

Dave MacKinnon, VP, product management, Clear-Com: From my experience managing multiple regional broadcast facilities, IP workflows have significantly enhanced intercom capabilities, allowing for high-fidelity audio and flexible routing. Our Arcadia Central Station’s recent upgrades further strengthen IP system scalability and centralized control with built-in redundancy to ensure mission-critical uptime.

Joyce Bente, president and CEO, the Americas, Riedel Communications: IP workflows using SMPTE ST 2110 have significantly improved intercom systems by providing ultra-low latency, high-fidelity audio, and built-in redundancy while simultaneously simplifying network architecture, so you don’t need different settings for video and audio systems. The standard’s network-based packet timing and stream separation allow broadcasters to instantaneously reroute audio over dual IP paths (e.g., via SMPTE 2022-7) to maintain seamless, synchronized communications — crucial during live events. Additionally, digital audio transport eliminates analog noise and supports echo cancellation and HD codecs, ensuring crystal-clear voice clarity even across distributed studio environments.

What integration challenges arise when connecting IP intercom systems with legacy analog equipment?

Adam Salkin, senior solutions architect, Diversified: Integration of IP intercom systems and legacy analog gear requires additional interface equipment such as MADI/2110, AES3/2110, Analog/2110. This creates tie-lines, with an associated need to manage them.

Martin Dyster, VP, business development, Telos Alliance: As intercom manufacturers move away from monolithic matrix systems, analog edge cards have been replaced with nodes (dedicated boxes) that convert baseband to IP. Where analog integration is needed, the use of nodes is considered more efficient than matrix cards because the node can be installed close to the source or destination, and the audio converted to IP locally, avoiding complex cabling infrastructure and potential noise interference associated with analog.

Dave MacKinnon, VP, product management, Clear-Com: During my time overseeing broadcast operations at NBC, I encountered challenges like balancing audio levels between analog and digital equipment, ensuring compatibility of control protocols such as GPIO and tally, and managing latency differences to maintain natural communication flow. Arcadia Central Station addresses these issues by providing seamless bridging between IP and legacy analog systems, simplifying integration and minimizing operational complexity.

Joyce Bente, president and CEO, the Americas, Riedel Communications: Overcoming these challenges typically requires careful architecture: bridging hardware, codec selection strategies, network upgrades (QoS, VLANs), and security design. While possible, integration demands both compatible formats, and design expertise to ensure smooth and feature-rich operation across mixed analog/IP installations.

How do production teams manage communication between remote sites and central studios using IP intercom solutions?

Roberto Musso, technical director, NDI: Production teams are adopting unified protocols that streamline the transport and control of multiple media types. Such protocols simplify the signal flow by encapsulating synchronized audio, video, and metadata over a single IP stream. This reduces the number of systems to configure and allows automation platforms to act more intelligently across the full production pipeline.

Martin Dyster, VP, business development, Telos Alliance: Transporting multicast AES67/ST2110-30 across WAN and/or the internet is not entirely impossible, but very difficult to achieve and can be unreliable. Consequently, manufacturers like Telos have created codec solutions like Infinity Link, which convert AoIP to VoIP to interconnect locations using protocols like OPUS and transport means such as NDI Audio and SRT. Dante Connect is becoming a popular option, too.

Dave MacKinnon, VP, product management, Clear-Com: Based on my broadcast leadership experience, production teams commonly use secure VPNs or dedicated WAN circuits to extend IP intercom connectivity between remote sites and central studios. Virtual panels and mobile apps empower crew and talent anywhere, while dual NICs and redundant routing ensure continuous communications. Arcadia Central Station’s IP routing capabilities and FreeSpeak Icon’s wireless mobility provide flexible, reliable communication across locations.

Joyce Bente, president and CEO, the Americas, Riedel Communications: Production teams can leverage secure VoIP via STUN/TURN and WebRTC to connect remote and central intercom endpoints, enabling seamless communication across different network environments without compromising performance. With Riedel’s Virtual Smart Panel, user management for secure access and authentication also makes endpoint configuration easy. These solutions use encrypted peer-to-peer connections (over TLS/SRTP) with STUN for firewall traversal and TURN for reliable relay backup, ensuring low-latency, high-quality voice even behind NATs or firewalls.

In what ways have broadcasters adjusted workflows to accommodate remote or cloud-based intercom platforms?

Roberto Musso, technical director, NDI: Broadcasters quickly learned that most audio-over-IP technologies weren’t built for cloud or public internet use, which created major gaps in remote production workflows. Intercom systems, traditionally hardware-based, had to change dramatically during the pandemic, forcing the industry to rapidly adopt software-based, decentralized solutions to maintain uninterrupted live production. Today, solutions like NDI are increasingly used not only for video and audio transport, but also for intercom communications, highlighting how essential and foundational it is to successful live production workflows.

Advertisement

Marc Wilson, managing director, Glensound: Broadcasters can connect commentary units to a PC or MAC using Dante Virtual Sound Card. Systems like Unity Intercom have become popular to provide multi-channel audio between the station and the commentary location. Commentary units can be remotely controlled using a separate IP address, while Unity handles bi-directional audio for commentary and talkback. 

Martin Dyster, VP, business development, Telos Alliance: Cloud-based production workflows evolved during Covid and have begun to mature to fulfill the accelerating need for content, particularly in live sports and news. Cloud video production systems are virtual versions of their hardware equivalents, and cloud intercom is no different. Our Infinity VIP users welcomed the flexibility of virtual comms and have adapted to software by using phones, tablets, and computers as intercom panels, in conjunction with Stream Decks as controllers where needed.

Dave MacKinnon, VP, product management, Clear-Com: Broadcasters I’ve worked with have embraced hybrid workflows combining on-premises hardware with cloud-enabled and remote intercom panels. Operating procedures now include clear handoffs between local IFB and remote talkback channels, and software-driven control rooms integrate communications into larger collaboration platforms. Clear-Com’s IP-enabled solutions support these evolving workflows without sacrificing audio quality or reliability.

Joyce Bente, president and CEO, the Americas, Riedel Communications: By combining AES67 Syntonous Mode (AES67 audio that does not require PTP sync), virtual smart-panel apps over secure VoIP, and redundant VPN/IP architectures, broadcasters have reengineered workflows to support seamless, high-quality, low-latency communication between remote sites and central studios — fully embracing distributed, modern production environments.

What considerations are critical when planning bandwidth and network architecture for IP-based comms systems?

Martin Dyster, VP, business development, Telos Alliance: AES67/ST2110-compliant networks require several important considerations. Firstly, the switches must support PTPV2, IGMP, and QoS as a minimum. For most AoIP audio systems, the network bandwidth is likely to be gigabit or above, although some legacy equipment may be 100MB, which must be allowed for. Consider the total signal throughput when selecting network hardware and choose accordingly. Many devices can run on PoE or PoE+, so also consider this when buying switches and ensure that the overall power draw from the equipment will not exceed what the switch can deliver.

Dave MacKinnon, VP, product management, Clear-Com: From a broadcast technology perspective, it’s essential to allocate dedicated bandwidth or use VLAN segmentation to avoid congestion and packet loss. Multicast traffic optimizes network efficiency when communicating with multiple endpoints. Redundancy and failover are critical to maintaining uptime during live productions. Arcadia Central Station offers robust network monitoring and management tools that help operators maintain optimal performance and respond quickly to issues.

How is audio mixing evolving?

Roberto Musso, technical director, NDI: The transition from hardware to software-based audio mixing is well underway, and adoption is accelerating across the industry. Software solutions offer unmatched scalability, flexibility, and the ability to evolve through updates rather than hardware replacements, offering advantages that traditional consoles can’t match. When combined with audio-over-IP technologies, these tools are redefining how audio is managed in modern production environments.

What are the essential features broadcasters are seeking… or should be seeking?

Chris Scheck, head of marketing content, Lawo: In addition to providing pristine audio quality, a broadcast-grade mixing console should allow engineers to do as much as possible directly from the sweet spot. Broadcasters furthermore expect an integrated platform where the main console can be complemented with a compact sidekick for grams and other contributions. Finally, the ability to prepare and monitor audio mixes in different delivery formats — various immersive flavors, 5.1 and stereo — simultaneously while staying in the creative flow, as well as intuitive on-screen channel routing are must-haves.

How has the adoption of IP audio standards, such as AES67, impacted audio mixing workflows?

Adam Salkin, senior solutions architect, Diversified: With a true 2110-based mixer, the audio mixer, audio router and intercom no longer need to be treated as three separate items. They can all have access to the same audio flows.

Chris Scheck, head of marketing content, Lawo: The Ravenna, AES67 and ST2110 suite of standards make all signals available anywhere on a network that can span several locations in different parts of the world, if necessary. These standards are WAN-compatible, and some of our customers leverage the fact that it is perfectly possible to have a mixing console in one location while the audio is processed elsewhere. “Permanent remote” workflows are emerging all over the world, with studios and one audio control room in one city and the processing tech stack in another.

Advertisement