VoIP call quality can vary significantly with different audio codecs, even with a good connection.
Audio codecs play a crucial role in determining the quality of Voice over Internet Protocol (VoIP) calls. They’re responsible for compressing and decompressing the audio signal to enable efficient transmission over the internet.
Choosing the correct audio codec for VoIP calls can significantly impact call quality, bandwidth utilization, and overall user experience.
In this post, we’ll discuss what audio codecs are, the different types of VoIP codecs, and factors that could impact your audio codec choices. Keep reading to learn more.
Audio codecs are software algorithms that compress and decompress digital audio signals.The primary purpose of an audio codec is to reduce the size of an audio file without degrading its quality.
This compression is essential for VoIP calls as it enables efficient audio data transmission over the internet. A codec on the receiving end then decompresses the compressed data to make it usable for the VoIP receiving device.
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VoIP codecs typically use lossy compression, which discards some audio data to compress it as much as possible. Discarding this little bit of audio data enables a lossy compression codec to reduce audio data to one-eighth or one-tenth of the original size, making VoIP calling more efficient.
Many VoIP codecs use sampling to determine which audio data can be safely left out. With this method, the codec collects a certain number of audio samples per second and discards the least important audio data. Then, when the codec compresses audio data, it uses these samples to determine which sounds can be discarded with the most minimal impact on audio quality for the end user.
The codec leaves out more data during lossy compression to achieve higher compression ratios. That’s why compressing the audio data to one-eighth of the original size produces better audio quality than compressing the audio data to one-sixteenth of the original size. It’s also why you need more bandwidth to handle higher-quality VoIP calls.
Generally speaking, the more your VoIP codec compresses the audio data, the more audio quality you’ll lose. But even with lossy compression, you can achieve very high-quality VoIP call audio. The key is that your VoIP codec carefully selects which audio data gets discarded during compression and doesn’t discard too much audio data.
If your VoIP codec discards too much audio data or does a poor job of selecting which audio data can be safely left out, you’ll get grainy, distorted voice calls. In addition, too much discarding can have a negative impact on any business where negotiation, sales, or customer service are handled over the phone.
Ultimately, you want to select a VoIP codec that reduces your bandwidth requirements as much as possible while retaining clear voice quality.
When it comes to selecting the right audio codec for your business’s needs, there are several types to choose from in VoIP technology. Each comes with its own set of tradeoffs, which we’ll discuss below. The most common VoIP codecs include:
G.711 is an uncompressed codec that produces high-quality audio but requires a high bandwidth connection. G.711 is widely used in traditional phone networks and is compatible with most VoIP devices.
This codec requires at least 96Kbps of bandwidth per line. If you want slightly better audio quality, you can use this codec with less compression. However, that will require 112Kbps or 128Kbps of bandwidth per line.
The G.711 codec is a good choice for your business’s VoIP calling needs if you require high-definition audio quality. It’s also a good option if your VoIP infrastructure requires connecting to the Public Switched Telephone Network (PSTN) since the G.711 codec uses no digital compression.
If you send faxes through your VoIP provider, you should use the G.711 codec because compression will cause faxes to fail. Since G.711 uses no digital compression, you’ll be able to send faxes through your VoIP carrier’s IP network.
The G.722 codec is one of the most versatile codecs. At the highest compression rate, this codec requires only 32Kbps per line. However, if you have more bandwidth and want better audio quality, you can use up to 128Kbps per line.
This codec works well if you want the option to increase compression to temporarily add more VoIP lines using the same amount of bandwidth.
G.729 is a compressed codec that produces lower-quality audio but requires less bandwidth than G.711. G.729 is commonly used in VoIP systems to save on bandwidth utilization.
At the highest compression rate, this codec requires just 12.8Kbps per line. However, it typically requires 16Kbps or 23.6Kbps per line.
The G.729 codec’s audio quality is quite good, especially considering its low bandwidth requirements. So this codec is a good choice for your business’s VoIP calling if you need to connect high volumes of VoIP lines. The audio quality is high enough for business calls, and the bandwidth requirements are low enough to make efficient use of your internet connection.
A modern, open-source codec, Opus produces high-quality audio while maintaining low bandwidth requirements. As a result, Opus is becoming increasingly popular in VoIP technology due to its superior performance in varying network conditions.
Developed by Skype, SILK is designed to deliver high-quality audio with low latency. SILK is used in Skype and other Microsoft products and is known for its performance in unstable network conditions.
iLBC is a codec designed for use in low-bandwidth networks. iLBC produces lower-quality audio but is still intelligible, making it an excellent choice for VoIP calls in remote or rural areas.
Choosing the right audio codec for VoIP calls is crucial to ensuring high-quality call performance. The codec used can impact several factors, as discussed below.
Your chosen audio codec can significantly impact the quality of your VoIP call. Uncompressed codecs, such as G.711, produce high-quality audio. However, they require high bandwidth connections, which may not be suitable for all users. On the other hand, compressed codecs, such as G.729, produce lower-quality audio but use less bandwidth.
Your audio codec can also affect the amount of bandwidth used during a call. Using a codec that requires less bandwidth, such as G.729 or Opus, can help save on bandwidth utilization, allowing for more efficient audio data transmission.
Latency is the amount of time it takes for an audio signal to travel from one endpoint to another.
The audio codec you use can impact the delay between when a user speaks and when the other party hears it.
Codecs with low latency, such as SILK, can help improve call quality in unstable network conditions. High latency can cause delays in conversation, making it difficult for callers to have a natural conversation. Some codecs are designed to have low latency, making them a better choice for VoIP calls where real-time communication is essential.
Choosing the correct audio codec for your business’s VoIP calls will always require some tradeoffs between call quality, bandwidth, and network capabilities. Below, we’ll look at some of the principal factors you should consider when selecting the right audio codec for your organization.
One of the most significant tradeoffs when selecting an audio codec for VoIP calls is the balance between audio quality and bandwidth usage.
In general, the higher the audio quality, the more bandwidth the codec will require. However, codecs that use more bandwidth can cause network congestion, even if the audio quality is high. Conversely, codecs that use less bandwidth may not produce the best audio quality, but they can help ensure calls are more reliable and less likely to be affected by network congestion.
The quality of a VoIP call can be affected by various network conditions, including packet loss, latency, and jitter. Some codecs are designed to perform better in unstable network conditions, making them a better choice for VoIP calls in environments where network quality is unpredictable.
Different codecs may be compatible with certain hardware and software platforms. Selecting a codec that works with your VoIP system is essential to ensure you get the best possible call quality.
A weak VoIP codec could cost you big bucks if your organization conducts business over voice calls. Therefore, before you select an audio codec, it’s important to know how much bandwidth you have and how much you’ll need. That way, you can avoid using a codec with more compression than you need.
You can estimate how much bandwidth you need based on how many VoIP lines you have. You can use 115Kbps per VoIP line. However, that’s a very loose estimate that errs on the side of avoiding using more bandwidth than you have.
For a more specific calculation, use this VoIP bandwidth calculator to choose which audio codec you plan to use. VoIP calls require less than 0.5Mbps of bandwidth for a single line, regardless of which codec you use. Based on the rough estimate of 115Kbps of bandwidth per line, you could support a few VoIP lines with only 0.5Mbps of bandwidth.
However, it’s essential to remember that you’ll need two channels per VoIP line—one inbound and one outbound. Lastly, it’s wise not to use 100% of your total bandwidth. Instead, you should only use about 80% of your total bandwidth to leave room for variances in network performance.
A more specific bandwidth requirement calculation will enable you to use your internet connection more cost-efficiently for VoIP calling. It also saves you from having insufficient bandwidth for all the VoIP lines you need.
If you have more VoIP lines than your internet connection can handle, you’ll experience a lot of jitter, distortion, latency delays, and even dropped VoIP calls. When your VoIP network has more traffic than it can handle, it must prioritize that traffic. As a result, some data will get dropped because there’s simply not enough room on the network for all that data.
That’s why it’s crucial to work out your bandwidth requirements and ensure your internet connection is capable enough for all your VoIP lines. Otherwise, your VoIP calling will be unreliable and unusable for business calls.
We’ve given you quite a bit of information to absorb about VoIP codecs—and somehow, we still have more ground to cover. Below, we’ll address some frequently asked questions about VoIP codecs to round out this high-level look.
Yes, as long as your VoIP provider enables you to select which codec you use, you can set your VoIP codec and make calls to test the audio quality.
The G.729 codec is the most commonly used VoIP codec because it has an excellent balance of audio quality and low bandwidth requirements.
The G.729 codec uses a high-complexity algorithm for audio compression. The G.729a codec uses a less complex algorithm and requires less processing power. However, the lower complexity algorithm also produces slightly lower audio quality.
You’ll experience call quality issues if your internet connection’s latency exceeds 250 milliseconds, regardless of which VoIP codec you use. The G.729 codec works well with high latency connections because it requires relatively little bandwidth, which reduces network stress.
Choosing the right audio codec for VoIP calls is critical for ensuring high-quality call performance. Your codec can impact call quality, bandwidth utilization, and latency. Organizations should evaluate their network requirements and select an audio codec that meets their needs, taking into account factors such as bandwidth, network conditions, hardware and software compatibility, and latency.
By selecting the right codec, organizations can ensure that their VoIP calls are reliable, high-quality, and cost-effective. However, it’s important to note that your VoIP carrier may determine your VoIP codec. So be sure to ask your VoIP provider about their available audio codecs and whether they offer VoIP codec selection.
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