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:
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Bit depth – the resolution of the audio signal
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Sample rate – the number of samples per second
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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:
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Protocol headers
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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.
🎧 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:
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Bitrate
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Audio quality
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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:
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Latency settings
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Buffer size
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FEC protection level
Typical protocol overhead (indicative):
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Pro-MPEG FEC: 20 – 125 % (deterministic, depending on FEC matrix configuration)
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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
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When configuring PCM streams, the resulting bitrate is automatically calculated based on bit depth, sample rate, and channel count.
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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:
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Net bitrate of the audio encoder
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Actual transmission bitrate including protocol overhead
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Protocols in use and packet loss statistics

On the Decoder Side
The overview page provides detailed streaming statistics, including:
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Net and streaming bitrates
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Missed packets and packet error rate (PER)
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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.