Articles – 2wcom

A Real Deadline, Not a Demo: Radio Training Network’s MoIN Deployment Story

UBy Dns — Thu, 12 Mar 2026 11:18:07 +0000

When Radio Training Network reached out about additional licensing, it was not part of a routine expansion. A real operational deadline had just landed on the table.

Late on a Thursday night, the team learned they had only ten days to absorb one of their sister networks and migrate its broadcast operations into their Riverview, Florida facility. Multiple full-power stations and translators across several states needed to be integrated, covering two different formats, all without any service disruption.

According to Aaron Schultz, IT Director and Assistant Engineer at Radio Training Network, the first questions from management were simple and direct. What would it cost, and could it be done in time?

The answer came down to flexibility.

Rather than deploying new physical STL hardware or scheduling site visits under heavy time pressure, the team chose a software-driven approach. Using MoIN as the transport layer and existing IP-4c units at transmitter sites, a new MoIN pod was spun up and SRT paths were established back to the studio and out to each site for both primary and failover audio.

The result was immediate.

In under 30 minutes, the MoIN pod was live, all audio paths were established, end-to-end audio was verified, and link stability was confirmed. No new hardware was installed at the studio, and no transmitter site visits were required.

Under the previous box-to-box model, this type of deployment would have meant additional STL hardware, physical integration work, and more potential points of failure. With MoIN, the entire setup was handled in software.

As the team approached final cutover, the transition itself was straightforward. Live program audio could be rerouted from the Riverview studios to transmitter sites across multiple states remotely with just a few clicks. The entire process was controlled centrally and executed with confidence.

For Radio Training Network, this deployment further validated a strategy that has been in progress for the past two years. The combination of MoIN as a software transport platform and IP-4c hardware at transmitter sites has enabled faster deployments, lower capital expenditure, fewer physical dependencies, and significantly greater operational agility.

This was not a lab test or a planned demo. It was a real deadline, real stations, and real operational pressure. The fact that it was executed entirely from desks, without emergency site work or last-minute hardware installations, highlighted the maturity and practicality of the platform.

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hsf Student Radio and our IP-4c

Melina Preuss — Thu, 22 Jan 2026 10:49:05 +0000

A little while ago, around the end of last year, we had the pleasure of supporting hsf Studentenradio e.V. with a special anniversary gift: an IP-4c audio codec, provided on the occasion of their 75th anniversary.

Since then, hsf has already shared their own experience and impressions in a LinkedIn post, which we were very happy to see and appreciate greatly.

The IP-4c was supplied to help improve the audio link between the studio and the transmitter site at Vogelherd. The existing connection, based on a shared IP microwave link, is prone to occasional packet loss — a challenge the IP-4c addresses with features such as FEC, RIST, and SRT, resulting in a more stable and robust transmission.

Additional features like detailed packet loss statistics support ongoing link optimization, while dual streaming enables redundancy via a separate network. In the event of a failure, the integrated backup switching allows automatic fallback to a web stream or file playback.

Even though the Christmas lights are long gone, we’re still very happy to have supported hsf Studentenradio e.V. and wish them continued success in their anniversary year and beyond.

How HAMBURG OPEN Brought Us Together

Melina Preuss — Thu, 08 Jan 2026 08:54:08 +0000

Just in time for this year’s HAMBURG OPEN, we wanted to share a little story from behind the scenes and how Bjarke found his way to 2wcom.

Bjarke Wrobel joined the 2wcom team not too long ago and now works with us as a Global Sales Engineer. When we first met him at HAMBURG OPEN 2025, he was still working as a Broadcast Engineer and giving a talk about his university project “Remote Jack”, focused on remote production.

During the event, Bjarke noticed 2wcom Systems as one of the companies from his hometown, Flensburg, and decided to stop by for a quick chat. As it turns out, our COO Mirko Goldschmidt had attended Bjarke’s talk earlier that day, which made the conversation even more interesting.

One chat turned into a longer one, everything clicked, and the rest pretty much wrote itself.

Since then, Bjarke has become a familiar face at 2wcom and will be back at HAMBURG OPEN this year, representing 2wcom Systems together with the rest of the team.

Whether you are looking for a new challenge, want to reconnect, or just want to learn more about our latest broadcasting technologies, we would love to see both familiar and new faces at HAMBURG OPEN 2026.

Find us on January 14 and 15 at Hamburg Messe and Congress, Hall 6, Booth B6.216 and say hi!

Not Just the Headliners – What Else We’ve Been Up To

UBy Dns — Wed, 10 Sep 2025 10:35:01 +0000

What's New at 2wcom

We promised real-world solutions instead of static screens, and this year, we’re showing how much progress we’ve made. As described in our previous article about highlights and demos, we focus on real setups. From cloud-native workflows to advanced transmission tools, our portfolio has grown with many new features and products ready for hands-on testing at booth 8.D81.

Expanding MoIN

MoIN, our flexible software-based AoIP core for managing contribution, distribution, and monitoring workflows, has seen a remarkable evolution. Beyond its flexible role in AoIP environments, it now integrates Stereo Tool sound processing for professional on-air audio quality. A new audio mixer allows combining multiple sources directly inside MoIN, while the SIP Call Manager provides straightforward control over multiple SIP connections.

MoIN now supports µMPX encoding/decoding, strengthening workflows where cost-efficient FM distribution is key. For broadcasters looking at multi-platform contribution, the addition of DAB+ encoding is a milestone – MoIN can now deliver DAB subchannels directly to multiplexers using EDI or Fraunhofer MuxEnc.

DAB+ Innovations

Since the release of the DAB+ encoder in MoIN and the IP-8e DAB edition, we’ve expanded with mux compatibility across vendors, a DAB sub-multiplex editor, and local PAD (DLS/SLS) import.

To strengthen robustness, we added SRT encapsulation for the feed from DAB+ encoder to the head-end, ensuring reliable delivery to multiplexers even over public networks.

For DAB distribution, we’ve enhanced our DAB-4c and DAB-4s with expert timing controls for all EDI sources. While SRT improves resilience, the new expert settings add another layer of protection: offset monitoring, alarms, and automatic muting ensure SFN stability and reliable DAB distribution. This gives operators greater confidence and peace of mind in maintaining seamless service.

FM and MPX Innovations

For FM distribution, the migration to µMPX has gained momentum, backed by new products like the MPX-2d. To support these workflows, the MPX Spectrum Analyzer provides real-time signal visibility with zoom levels, markers, and export options.

Our solutions now support FM/HD alignment, guaranteeing a constant relative delay between analog FM and HD Radio signals.

Introducing the newest member of our MPX family: the MPX-2d FM-MPX over IP Decoder – a lean, cost-effective solution for reliable IP-based MPX decoding. Designed for broadcasters who value efficiency, this decoder supports up to two channels, making it scalable to fit networks of all sizes.

Planning to transition from audio to MPX? The MPX-2d and MPX-2ds offer the flexibility to decode both MPX and audio feeds, ensuring seamless integration into your workflow regardless of your current setup.

We’ve also introduced the MPX-1g Alarm Receiver, which combines STL codec, stereo encoder, RDS, and alarm functions in a single unit. When PTY31 is activated in a national FM transmission, the MPX-1g automatically switches from the local STL feed (via µMPX, MPX, or audio) to the national broadcast. This ensures listeners receive critical alerts without delay.

nettwark– Device Control Reinvented

With nettwark, we’re redefining how broadcast devices are managed and controlled remotely. As the modern, web-based successor to our discontinued SIRC, nettwark offers an intuitive and efficient way to handle device management. With its modern Linux-based architecture, optimized IP transport mechanisms, and seamless integration – even over satellite links – nettwark streamlines operations, reduces effort, and ensures reliable control across any network.

Remote Contribution

With up2talk, we’re demonstrating live sports commentary workflows featuring low-latency audio contribution alongside video in/out. A new talkback function ensures seamless coordination in live production.

Audio and Stream Monitoring

On the monitoring side, the eyera platform provides a live status overview of audio streams. Supporting a variety of audio sources – including EDI for DAB and µMPX for FM – eyera is a powerful and audio-centric tool to ensure quality in audio workflows. Complemented by stream statistics, alarming, and loudness measurement, operators can rely on clean, spec-compliant delivery.

Platform Extensions – Flexible Software and Modular Hardware

Loudness measurement according to EBU/ITU recommendations and extended logging add confidence in compliance and troubleshooting.

Our solutions now embed SCTE-35 splicing markers and support decoding for automation, opening up new opportunities for advertising insertion and regional opt-outs. In addition, we have integrated Kantar watermark decoding, enabling customers to monitor the presence and confidence of watermarks throughout the distribution chain.

New hardware options include a DVB-S2X dual tuner for SAT-4d, SAT-4da, DAB-4s and MPX-2ds, future-proofing satellite contribution.

Come by booth 8.D81 at IBC!

From DAB+ encoding to SIP call management and MPX spectrum analysis, the progress we’ve made since last year reflects one goal: making broadcast contribution and distribution reliable, flexible, and future-ready.

How Broadcasters Bring Remote Voices Into the Game with up2talk

UBy Dns — Thu, 21 Aug 2025 11:47:05 +0000

Bringing an outside guest or commentator into a live TV program should be simple. In practice, it is often a challenge: unstable connections, poor audio quality, or complicated setup steps that create stress for both guest and production team.

up2talk was designed to solve this. Built on broadcast standards and refined through close collaboration with broadcasters, it offers a barrier-free way to connect external voices with studio-level quality.

The Challenge

A leading live sports production needed to seamlessly integrate remote commentators and experts into their broadcast. The requirements were clear:

Moderator must be able to join with a single browser link, no software installation, no training.
The live TV program should be streamed back to the guest in real time, ensuring their commentary is perfectly aligned.
Audio and video must meet the reliability and quality expectations of broadcast production.

Our Solution with up2talk

Simple Guest Access: One link, one browser, guests join instantly via WebRTC.
Program Feed Return: The ongoing program is streamed back into the guest’s browser, keeping them in sync with the live broadcast.
High-Quality Transport: Audio and video are transmitted via SRT in H.264/AAC, ensuring robust performance across networks.
Talkback Channel: A dedicated push-to-command path allows direct coordination between the guest and the production control room.

The Result

Remote commentators and experts are brought into the production with clarity and reliability. The audience experiences a seamless broadcast, while production teams gain confidence and efficiency in handling external voices.

Experience up2talk at IBC in Amsterdam

This is just one example of how up2talk is redefining remote contribution for broadcasters. Curious to see it in action? Visit us at IBC 2025 in Amsterdam and explore our live demo at the 2wcom booth.

IBC Amsterdam
12. September – 15. September 2025
2wcom Booth: Hall 8, D81

Book your up2talk Demo at our booth now – Click below!

Is DAB+ HE-AAC? Not Quite.

UBy Dns — Thu, 31 Jul 2025 11:14:30 +0000

Why audio encoding for DAB+ requires more than just picking a codec profile.

DAB+ uses the HE-AAC v2 profile, the same one used in MPEG-4 applications like streaming and television broadcasting. So why isn’t it just “plug-and-play”?

In reality, integrating HE-AAC v2 into a DAB+ system requires strict adaptations. From frame structures and bitrate caps to reconfiguration constraints and encoder behavior, DAB+ brings its own set of rules.

This article breaks down what makes DAB+ encoding unique and why encoder behavior plays a critical role in delivering consistent audio quality.

DAB+ is just HE-AAC, isn’t it?

DAB+ audio encoding uses the HE-AAC v2 profile (High-Efficiency Advanced Audio Coding version 2), a standard also found in MPEG-4 applications such as television broadcasting and streaming. However, to integrate this codec within the Digital Audio Broadcasting (DAB+) system architecture, several specific adaptations and restrictions were introduced. These changes are crucial for synchronization, reconfiguration, and overall system compatibility.

Decoders Are Ready – Encoders Decide

HE-AAC v2 is formally a decoder profile, defined in the MPEG-4 standard. HE-AAC v2 is a superset of AAC-LC and HE-AAC. This means that a HE-AAC v2 decoder is always backward compatible and can correctly decode:

AAC-LC (Low Complexity) only
AAC-LC + SBR (equivalent to HE-AAC v1 profile)
AAC-LC + SBR + PS (equivalent to HE-AAC v2 profile)

Whether SBR (Spectral Band Replication) and PS (Parametric Stereo) are used depends entirely on the encoder configuration, not the decoder. For example, for bitrates below 96 kbit/s the usage of SBR usually provides a quality benefit compared to using AAC-LC without SBR.

System Constraints in DAB+

To integrate HE-AAC v2 cleanly into the DAB system, a number of constraints were applied. These include:

A maximum subchannel bitrate of 192 kbit/s
Support for only two audio sampling rates: 48 kHz or 32 kHz
A DAB frame length of 24 ms, which differs from native AAC frame durations

These limitations not only define the boundaries of what the codec can do within DAB+, but also drive the need for deeper structural changes in encoding.

Frame Length and Audio Super Framing

A key challenge in adapting AAC to DAB+ lies in frame alignment. A frame represents a short chunk of audio that the encoder processes at once. AAC-LC usually processes 1024 audio samples per frame, corresponding to about 21.3 ms, while HE-AAC and HE-AAC-v2 (which includes SBR) use 2048 samples per frame – approximately 42.6 ms. Neither fits well with DAB’s fixed 24 ms frame structure.

To resolve this mismatch, DAB+ mandates the use of shorter frame lengths:

960 samples per frame for AAC-LC (20 ms)
1920 samples per frame for HE-AAC and HE-AAC v2 (40 ms).

These adapted frame lengths are then grouped into audio super frames to match the transmission structure. Each super frame has a duration of 120 ms, regardless of the codec profile used.

Depending on the sampling rate and whether AAC-LC or HE-AAC is used, the number of audio frames per super frame varies. For example:

HE-AAC at 48 kHz sampling rate groups 3 frames per super frame.
AAC-LC at 48 kHz groups 6 frames of 20 ms each into a super frame.

These super frames are distributed across five DAB frames, enabling time-aligned, frame-exact transmission.

Why Super Frames Must Be Self-Contained

In traditional AAC encoding, bits can be shifted between adjacent frames to better represent complex audio content. However, DAB+ forbids such inter-frame dependencies. Each super frame must be 100% self-contained to support:

Seamless switching between services or configurations
Error resilience
Clean reconfiguration boundaries.

This design is critical in broadcast environments where dynamic reconfiguration or service-following is required—for example, when users switch stations or when broadcasters adapt content on the fly.

Trade-Offs and the Need for Encoder Tuning

These structural adaptations – shorter frame lengths and rigid super framing – do affect coding efficiency. The encoder has less room to manage bursts of audio complexity, and fewer tools to smooth over transitions. As a result, achieving high audio quality, especially at low bitrates, requires careful encoder optimization.

The 2wcom Approach

At 2wcom, we place special emphasis on audio fidelity and system compatibility. Our DAB+ encoders – featured in the IP-4c, IP-8e, and MoIN platforms – are built around the Fraunhofer Professional DAB+ Encoder library. This ensures:

Best-in-class audio quality, verified through extensive listening tests,
Full compliance with the DAB+ system specifications
Reliable performance under real-world broadcasting conditions

At 2wcom, we understand that delivering consistent audio quality in DAB+ environments requires more than just using the codec – it demands expertise in how to apply it. By building on the Fraunhofer Professional DAB+ Encoder library and focusing on encoder behavior under real-world conditions, we ensure that our solutions provide stability, synchronicity and adaptability. Whether broadcasters are operating at high or constrained bitrates, our systems are designed to maintain performance without compromising on audio integrity.

Codec Considerations: Choosing the Right Audio Format for Broadcast & Streaming

UBy Dns — Thu, 17 Jul 2025 11:17:05 +0000

This year marks the 30th birthday of the MP3 format. Back in 1995, scientists at Fraunhofer IIS in Erlangen chose the file ending .mp3 for MPEG-1 Layer 3 — and unknowingly changed audio consumption forever.

What many don’t realize: broadcasting was one of MP3’s first real-world applications.

Today, codecs like AAC, Opus, and xHE-AAC have evolved far beyond what MP3 offered — but its legacy lives on. So the question remains:

Which audio codec should I use in my workflow today?

Audio codecs are essential in modern media workflows – from digital broadcasting and livestreaming to remote contribution and archiving. But audio codecs differ significantly in performance, latency and efficiency. Selecting the right one depends on your needs: quality, latency, bandwidth, or compatibility.

What is Audio Coding?

Before choosing a codec, it helps to understand how raw audio (like PCM) becomes manageable for transmission and storage. A stereo 48 kHz/16-bit signal generates 1.5 Mbit/s – about 691 MB per hour. Multiply that by a thousand listeners, and the need for compression becomes obvious.

There are two types: lossless and lossy. Which one fits best depends on your use case:

Lossless (e.g. FLAC)	Lossy (e.g. MP3, AAC)
No audio information is lost – great for archiving and mastering	Uses perceptual models to discard inaudible details for better compression

Typical audio data rates for 1 hour of stereo audio:

Uncompressed PCM

FLAC (Lossless)

MP3 / AAC / Opus

≈1.5 Mbit/s

~700–1100 kbit/s

~96–256 kbit/s

Audio Coding Technologies

Different codecs are optimized for different tasks — from speech to music, from low latency to maximum efficiency. Here’s a quick overview of common codec families.

Speech Codecs

Examples: G.711, G.722, SILK
Low bitrate, low latency – optimized for efficient transmission of human speech.

Perceptual Codecs

Examples: MP3, AAC, Vorbis
Use psychoacoustic models to discard sounds that are less likely to be perceived, achieving compression without hurting audio quality.

Predictive Codecs

Examples: ADPCM, SILK, aptX
Work by encoding only the difference between audio samples. This results in lower CPU usage and stable quality

Adaptive Codecs

Examples: Opus, xHE-AAC
Can dynamically adjust bitrate and behavior depending on content and network conditions.

Perceptual Audio Coding

Not all sounds are heard equally, and perceptual audio coding takes full advantage of that fact.

Perceptual coding relies on how we hear: louder sounds mask nearby quiet ones — both in time and in frequency. These masking effects allow encoders to discard audio information that would be inaudible to human ears, saving bits without significantly harming perceived quality.

The most famous codec based on this principle is MPEG-1 Layer 3 — better known as MP3.

It revolutionized the way we consume audio and remains one of the most widespread formats to this day. However, MP3 is now outperformed by newer codecs like Opus and the AAC family.
At the same bitrate, these formats offer better audio quality — or to put it another way: modern codecs achieve comparable quality at lower bitrates than MP3. See PDF

Codec Delay – Why Latency Matters

When choosing a codec, bitrate and audio quality often take center stage — but latency can be just as critical.

In live broadcasts, remote contribution, or intercom systems, even a small delay can disrupt timing, interaction, and listener experience. Some codecs, like Opus or AAC-LD / AAC-ELD, are optimized for low-latency streaming. Others, such as HE-AAC v2, prioritize compression efficiency — but introduce more delay as a trade-off.

Codec	Delay	Efficiency
AAC-ELD	Low (~20–32 ms)	High
Opus	Very Low (~5–30 ms)	High
Enhanced aptX	Very Low (<5 ms)	Very High
FLAC	Low (~5–10 ms)	Lossless (heavy)

Which Codec for Which Application?

The best codec depends on what you’re trying to achieve — low delay, high quality, mobile efficiency, or infrastructure compatibility.

Here’s a practical overview based on common broadcast and streaming scenarios.

Use Case	Recommended Codecs	Why
Remote contribution, intercom, live-link	AAC-ELD, Opus, Enhanced aptX	Low latency
Mobile streaming, backup transmitter feeds	HE-AAC v2, xHE-AAC, Opus	Good quality at low bitrates
Studio-to-transmitter link (STL), DVB-Backbone	MPEG-1 Layer 2, AAC-LC, Enhanced aptX	Infrastructure compatibility
Music streaming, archiving	AAC-LC, FLAC, Opus (fullband)	High audio quality
Standard-compliant broadcasting (DAB+, DVB, DRM)	MPEG-1 Layer 2, HE-AAC v2, xHE-AAC	Required by standards
Adaptive streaming (HLS, DASH, mobile)	xHE-AAC, Opus	Dynamic bitrate switching

Built-In Quality: No Codec Compromises

MPEG specifies the bitstream format and decoder — but encoder quality can vary.

Our products use the Fraunhofer professional codec libraries for MPEG codecs, including MPEG Layer 2 and 3, as well as the complete AAC family.

This ensures high performance and superior audio quality across all supported workflows.

Isn’t DAB+ just HE-AAC?

DAB+ uses the HE-AAC v2 profile, but it’s not identical to what’s used in typical streaming workflows.

Specific adaptations have been made to meet the transmission and error protection requirements of digital radio systems.

We’ll explore the technical differences in an upcoming article.

Want to Dive Deeper?

Here’s a downloadable summary of the core codecs used in today’s broadcast and streaming environments — all of which are supported across the 2wcom portfolio:

Understanding Bitrates and Overhead in Audio-Over-IP Streaming

UBy Dns — Tue, 27 May 2025 12:39:19 +0000

When working with professional Audio-over-IP (AoIP) solutions, it’s crucial to understand how different audio formats and protocols impact your network’s data rate. In this article, we’ll break down typical data rates for various audio formats, explain the overhead introduced by streaming and robustness technologies like FEC, RIST, and SRT, and show you how to verify these parameters directly in the 2wcom codec GUI.

Typical PCM Bitrates – Just the Math

Uncompressed digital audio in Pulse-Code Modulation (PCM) format is defined by three key parameters:

Bit depth – the resolution of the audio signal
Sample rate – the number of samples per second
Channels – mono, stereo, or multi-channel

To calculate the net bitrate of a PCM audio stream, it’s as simple as:

Bitrate = Bit Depth × Sample Rate × Channels

Here are some common examples:

Format	Net Data Rate	Estimated RTP Data Rate (incl. overhead)
16 bit @ 48 kHz stereo	1.536 Mbit/s	~1.6–2.0 Mbit/s
24 bit @ 48 kHz stereo	2.304 Mbit/s	~2.4–2.8 Mbit/s
16 bit @ 192 kHz mono (MPX)	3.072 Mbit/s	~3.3–3.8 Mbit/s
24 bit @ 192 kHz mono (MPX)	4.608 Mbit/s	~5.0–5.5 Mbit/s

The difference between the net bitrate and the actual transmission rate comes from the overhead introduced by RTP, UDP, and IP headers added per packet. This overhead depends on two main factors:

Protocol headers
Packetization interval – the number of audio samples per packet

Smaller packets (shorter intervals) mean higher overhead but potentially lower latency. So, choosing the right packet size is always a balance between latency, overhead, and error resilience.

“Low delay is great—but if your packets are too small, you might be wasting bandwidth. Finding the sweet spot is what smart AoIP setup is all about.”

Compressed Audio: Audio Quality vs. Bitrate

A general rule of thumb? Higher bitrate usually means better audio quality—but it’s not always that simple.

Audio quality doesn’t rely on bitrate alone. It also depends heavily on the efficiency of the codec algorithm being used. Every codec has its own “sweet spot” where quality, bitrate, and latency are well balanced.

So, when you’re choosing a codec, keep all three factors in mind:

Bitrate
Audio quality
Latency

We’ll dive deeper into codec comparisons and trade-offs in an upcoming article.

Protocol Overhead: SRT, RIST, FEC & More

When using robust streaming protocols like FEC, SRT, or RIST for STL (Studio-to-Transmitter Link) applications, additional overhead is introduced. The exact amount depends on several configuration factors, such as:

Latency settings
Buffer size
FEC protection level

Typical protocol overhead (indicative):

Pro-MPEG FEC: 20 – 125 % (deterministic, depending on FEC matrix configuration)
RIST / SRT: 1 – 15 % (adaptive, depending on retransmissions caused by network stability)

Keep in mind: These values are indicative. For exact figures—especially when planning critical STL links—we recommend testing AoIP links under real network conditions or contacting our support team.

Verifying Bitrate and Overhead in 2wcom Codecs

All bitrate and overhead parameters are clearly visible in the 2wcom web GUI—giving you full control and transparency during setup and operation.

On the Encoder Side

When configuring PCM streams, the resulting bitrate is automatically calculated based on bit depth, sample rate, and channel count.
For compressed audio (e.g. AAC, Opus, MPEG Layer II), the bitrate is set manually as part of the codec configuration.

Once the encoder and output streams are active, the overview page displays key information at a glance:

Net bitrate of the audio encoder
Actual transmission bitrate including protocol overhead
Protocols in use and packet loss statistics

On the Decoder Side

The overview page provides detailed streaming statistics, including:

Net and streaming bitrates
Missed packets and packet error rate (PER)
FEC overhead and RIST/SRT retransmission statistics

Conclusion

Understanding bitrates and overhead in AoIP systems is essential for optimizing performance and ensuring reliable audio streaming.

Whether you’re configuring PCM links or fine-tuning codec-based workflows, 2wcom gives you full transparency and control—from raw data rates to robust protocol configurations.

Ensuring Robust Streaming Over Unstable Networks

UBy Dns — Thu, 03 Apr 2025 11:57:35 +0000

In today’s rapidly evolving media landscape, reliable audio and video streaming is essential for broadcasters, enterprises, and content providers. However, streaming over unmanaged and unstable networks – such as public internet connections – poses significant challenges. Packet loss, jitter, and latency fluctuations can degrade the quality of real-time streams, calling for robust streaming solutions.

Forward Error Correction (FEC)

Forward Error Correction (FEC) is a proactive strategy where redundant data packets are transmitted alongside the original packets. If data loss occurs, the missing packets can be reconstructed at the receiver without needing retransmission. While FEC effectively mitigates packet loss, it comes with increased bandwidth overhead.

FEC works by using row and column packet calculations to repair the stream at the receiving end. A higher redundancy factor provides better error correction but increases the required bandwidth, making it crucial to find a balance between effectiveness and available network resources. FEC is particularly useful in multicast scenarios where retransmissions are not feasible.

Dual Streaming

Dual streaming enhances robustness by transmitting the same content over two independent network paths. This method can be implemented in different ways:

Using different networks: Sending duplicate streams over diverse network infrastructures (e.g., one over cellular and another over satellite) reduces the risk of complete transmission failure.
Time-offset redundancy: When both streams are sent over the same network, introducing a slight delay between the two streams allows the receiver to switch to the backup stream if packets are lost in the primary stream.

Dual streaming requires twice the bandwidth of a single stream, ideally distributed across separate networks for optimal redundancy.

Reliable Transport Protocols: SRT vs. RIST

Secure and reliable transport protocols offer additional mechanisms to address network-induced impairments. Two widely adopted protocols in the industry are Secure Reliable Transport (SRT) and Reliable Internet Stream Transport (RIST). How do SRT and RIST compare? Here‘s an overview:

Feature	SRT	RIST
Industry Support	Widely used in media workflows, especially for video	Less widely supported for video
Accessibility / Standardization	Proprietary solution, SDK available (risk of vendor lock-in)	Based on open standards
(Backwards) Compatibility	Different SDK versions sometimes incompatible	Complements RTP streaming via RTCP; receivers without RIST support still receive RTP streams
Security	Built-in AES encryption	Encryption must be added externally
Latency	Typically lower latency due to ARQ (Automatic Repeat reQuest) mechanism ~100 ms to 2 s, depending on setup	More flexible tuning depending on buffering and FEC settings. Can be tuned for either low-latency or high-reliability streaming
Multicast Support	Limited, experimental support, not widely used	Fully supports multicast via RTP/RTCP
Bandwidth needed	Increases with retransmissions (ARQ)	More efficient with selective retransmissions and optional FEC

Choosing the Right Solution

Selecting the right approach for robust streaming is not a one-size-fits-all decision – it depends on specific needs, infrastructure constraints, and industry requirements. Here’s how different use cases align with the available technologies:

Broadcasters may prefer SRT due to its widespread industry support and built-in security.
Organizations favoring open standards may choose RIST to avoid vendor lock-in and maintain backward compatibility with RTP-based streaming.
Mission-critical applications may implement a combination of dual streaming or FEC and reliable protocols to achieve the highest reliability.
Users needing multicast support should opt for RIST, as SRT does not natively support multicast.

As the demand for high-quality, uninterrupted streaming continues to grow, leveraging a combination of these strategies ensures robust and reliable content delivery – even over the most challenging network conditions.

Test Our Error Protection at NAB 2025

Do you want to see these error protection mechanisms in action? Visit our booth at NAB 2025! Our IP-4c device will be set up with a music playlist and four different error protection strategies: no protection, FEC, SRT, and RIST. Attendees will have the opportunity to switch between channels to compare audio quality under different protection methods, emulate different packet loss levels on the transmission to directly observe their impact on each stream, and explore the WebUI of our devices to see real-time data and adjustments.

Don’t miss this unique opportunity to experience these error protection methods firsthand!

Mastering Studio Transmitter Links (STLs) for Broadcast Excellence

UBy Dns — Tue, 26 Mar 2024 09:04:38 +0000

There are numerous considerations to address when configuring a Studio to Transmitter Link (STL) for your radio station. From what needs transporting to where it needs to go, and in what format it should arrive, each decision impacts the efficiency and effectiveness of your broadcasting setup.

The communication pathway between studio and transmitter site(s) often dictates subsequent decisions. Over the past decade, options for transporting audio, data, and other information have evolved significantly. Traditional TDM networks, once prevalent, have become impractical due to cost constraints, prompting a shift towards alternatives like the 950 MHz band for radio STLs.

However, challenges arise with 950 MHz systems, including saturation in many markets and limitations in bandwidth and functionality. While not obsolete, 950 STL systems may necessitate additional equipment for full functionality.

Today, IP-based circuits have emerged as the preferred medium for STLs, offering versatility across microwave, fiber optic, and copper connections. These circuits boast full duplex capabilities, enabling bidirectional transmission of audio and data—a crucial feature for remote site management and performance monitoring.

Crucially, a well-designed STL system ensures consistent throughput and minimizes performance issues such as latency, packet loss, and jitter. Understanding bandwidth requirements is paramount in selecting suitable equipment for your setup.

Audio transport options range from discrete signals to composite formats, each offering distinct advantages based on studio infrastructure and transmission needs. Leveraging advancements in IP circuit capacity and audio codecs, broadcasters can achieve optimal audio quality and minimal latency even with restricted bandwidth.

While Ethernet-based systems offer ample bandwidth, they pose challenges in live traffic transport due to packet routing variations. Specialized tools like Secure Reliable Transport (SRT) and Forward Error Correction (FEC) mitigate these issues, ensuring reliable signal delivery.

Furthermore, maintaining synchronization between analog and digital signals in applications like HD Radio demands precise timing control. Solutions like 2wcom’s MPE circuit facilitate seamless integration and alignment, simplifying the implementation and maintenance of HD Single Frequency Networks (SFN).

In essence, your STL serves as the conduit for delivering your meticulously crafted audio content. A well-optimized system not only preserves quality but also enhances the user experience, offering features like SFN. When it’s time to upgrade your STL, trust 2wcom’s expertise and support for a seamless transition. We’re here to help you design a trouble-free STL system tailored to your needs.

Innovation Crossroads: The Intersection of Software and Embedded Hardware

WPUser — Tue, 19 Mar 2024 13:13:38 +0000

In the ever-evolving landscape of technology, businesses and developers are constantly faced with a crucial decision: whether to opt for software solutions or rely on embedded hardware. This dilemma underscores the fundamental trade-offs between scalability, energy-efficiency and reliability. In this article, we delve into the comparison of these two approaches, examining their respective strengths, weaknesses, and real-world implications.

Scalability

In very general sense, software solutions provide better scalability than embedded hardware. Of course this has to be compared case by case, but with the right underlying infrastructure software solutions can leverage server clusters to easily scale and share ressources (e.g. using Kubernetes). For example our MoIN solution can startup a codec in a couple of seconds while adding a new hardware codec would require technicians to install the codec, attach cables, provision IP addresses and more.

Energy Efficiency (and in consequence C02 savings)

To reduce energy consumption as well as saving C02 equivalents (CO₂e) the required density per location is an important factor. In places where only a few audio channels need to be received, for example at radio transmitters, embedded hardware can outperform software solutions on server hardware by far. But in centralized head-ends where many channels are processed at the same time, software seems to be the clear winner. Even though dedicated hardware has a better efficiency than CPU based processing, it becomes clear very quickly that when you replace 400 hardware codecs by only 6 servers that the energy consumption is reduced significantly.

Reliability

As a company that both creates software solutions and embedded hardware we know that it is very difficult to foresee errors that occur on hardware that you don’t control such as standard server hardware or even PCs. So manufacturing the hardware and creating software for an embedded device helps with controlling every part of the processing and increases reliability. In contrast when it comes to a huge amount of channels redundancy plays a big role to increase availability and reliability of a system. Therefore software solutions and frameworks like Kubernetes can drastically improve the redundancy by using multiple replicas and even implement “self-healing” capabilities to restore the desired condition of a system.

TL;DR – Conclusion

The decision for embedded hardware or software solutions has to be evaluated case-by-case and unfortunately there is no general answer. We are living in a hybrid world in so many areas already. So we at 2wcom think that embedded hardware and software will co-exist and as a rule of thumb: the higher the density the more likely it will be that software solutions outpace embedded hardware.

Next Generation MPX Solutions

UBy Dns — Wed, 02 Aug 2023 09:24:53 +0000

If you are interested in the newest updates for MPX and µMPX in radio broadcasting, we are excited to share with you the latest advancements in high-quality products from 2wcom Systems GmbH.

We take pride in continuously pushing the boundaries of innovation in our MPX product line. Our dedication to meeting customer needs and incorporating valuable feedback has led to the development of two remarkable solutions: the MPX-2c and MPX-2ds. These cutting-edge offerings have already garnered attention and are now being implemented in major radio broadcasting projects worldwide.

One of the most noteworthy features in these new products is the introduction of Single Frequency Network (SFN) technology with 1PPS and Synchronous Playout Networks (SPN) using Network Time Protocol (NTP). By seamlessly synchronizing timing across multiple transmitters, these enhancements ensure optimal signal quality, coverage, and synchronicity. Broadcasters can now deliver a consistent and reliable experience to their listeners, no matter where they are tuning in from.

However, it is the MPX-2ds that truly steals the spotlight. This innovative addition tour MPX product range supports MPX over Satellite, making it an invaluable tool for broadcasters relying on satellite distribution. Acting as a satellite receiver, the MPX-2ds can output MPX signals transport via satellite, fulfilling the specific demands of customers who have eagerly awaited this capability. With this feature, broadcasters gain a new level of flexibility and accessibility in delivering their content.

Recognizing the growing importance of accurate timekeeping and synchronization in radio broadcasting, we at 2wcom have gone a step further by integrating a GPS module into our products. This addition ensures precise timing, facilitating seamless transitions between various broadcast sources and enabling uninterrupted operation in complex network environments. The integrated GPS module will be available in the newest versions of 2wcom MPX products, guaranteeing exceptional performance for broadcasters in the ever-evolving industry.

With these remarkable updates, 2wcom Systems GmbH continues to cement its position as a leading provider of advanced solutions for radio broadcasting. The MPX-2c and MPX-2ds, embody the company’s commitment to deliver products that exceed customer expectations.

To learn more about these updates and how they can elevate your broadcasting capabilities, get in touch with our knowledgeable sales team. Discover firsthand why broadcasters worldwide are embracing these advancements and how they can revolutionize your radio broadcasting experience.