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The Role of VoIP Codecs in the Call Audio Quality

The Role of VoIP Codecs in the Call Audio QualityThe Role of VoIP Codecs in the Call Audio Quality
How do VoIP codecs help call audio quality?

Updated: March 5, 2024

Voice over Internet Protocol (VoIP) is a modern protocol where communication occurs by transferring data packets over the Internet. Utilizing VoIP phone systems has several benefits over traditional telephony, driving its popularity in enterprises. 

VoIP provides lower calling rates, reducing the costs for businesses. Because it only requires the Internet, VoIP phone systems are very flexible and portable, which is great for team members working remotely and those who travel frequently. 

VoIP phone systems offer advanced features such as call forwarding and waiting. They also have integration capabilities, allowing seamless collaboration with other digital tools and software.

Despite all the advantages VoIP services provide, several problems can also affect them. These issues may result in choppy audio, echoes, and other poor audio quality issues. VoIP services use codecs to compress audio before sending and decompress after receiving. The compressing and decompressing can significantly affect audio quality, especially since different codecs provide different performance levels.

This blog post will explore how codecs impact call audio quality. We’ll look at popular codec options and share insights into how to go about selecting codecs. We’ll also explore the importance of call audio quality and the factors affecting this quality in VoIP calls.


Understanding VoIP Codecs

What are VoIP codecs, and why do they matter?

VoIP codecs encode and decode voice signals for transmitting voice over IP networks. The codecs achieve this in different steps, and the codec's performance at each stage will impact call quality.

The codec first converts an analog voice signal from the phone microphone into a digital signal. It then compresses or encodes the raw digital signal to optimize the signal for transmission. Many codecs include fundamental analysis and differential coding among the standard compression techniques. 

The encoded data is then chopped and packaged into small packets using a compression algorithm. These data packets are then transmitted over the network. The packet size affects jitter and delays, and the compression algorithm chooses an optimal packet size for efficient transmission.

When the data packets reach the recipient device, the codecs unpack them and return the digital voice information in the correct order. It then decodes the compressed voice data to deconstruct the original audio. The final step involves converting the digital signal into an analog one played out through the speaker.

Importance of Call Audio Quality

Call audio quality is critical to effective communication for individuals and enterprises. Latency delays, missing chunks or audio, and other audio quality issues can lead to misunderstandings. These misunderstandings will have a direct negative impact on team productivity. Similarly, these issues will negatively impact the user experience when communicating with clients and other stakeholders, which can ruin their overall perception of your brand.

Factors Influencing Call Audio Quality

What factors influence the quality of an audio call?

The factors that affect VoIP call audio quality are interconnected. The good news is that you can effectively resolve these issues simultaneously with some deliberate effort. Constant and attentive monitoring will help reveal what factors affect your audio quality so you can start strategizing on how to deal with each for a better user experience.

Some of the factors that influence the quality of an audio call include the following:

Network latency: Network latency throws off the delivery timing without necessarily degrading the audio quality. These network delays will have the other person speaking over you, or you will hear them talk at odd and irregular times.

Jitter: Jittery speech patterns during calls result from irregular data packets arriving at their destinations. You will likely miss some audio if the data packets arrive too quickly. On the other hand, you will hear bouts of silence if the packets arrive too late.

Packet loss: The network may drop data packets altogether if they get delayed or lost along their transmission journey. The network may also drop data packets if they have errors. When packet loss happens, you will have missing chunks of audio on the call.

Performance of VoIP equipment: From routers and modems to firewalls, the components that make up the VoIP infrastructure must work together in unison. Build quality, age, compatibility, and other factors directly impact audio quality.

There are several options to choose from when selecting VoIP codes. The most popular among these are highlighted below.

G.711: This codec has a 64kbps bitrate for a single path. It provides high-quality audio and doesn’t require licensing fees. On the other hand, G.711 involves a lot of bandwidth and doesn’t do as well as different codecs in supporting multiple calls.

G.722: This codec offers audio quality over various bandwidths and compression rates. G.722 operates at a higher bandwidth than the G.711 codec, with permissible bandwidths for this codec including 32, 48, 56, and 64 kbps. The G.722 has a 16 kHz sampling frequency, improving audio quality and increasing efficiency.

GSM: The Global System for Mobile Communication (GSM) codes are compatible with most VoIP devices and use a high compression ratio to deliver excellent audio quality. It is used in VoIP systems with GSM cellular compatibility. While GSM codecs can use up to 64 kbps of bandwidth, they typically use 13 kbps or lower, with negligible to no sound quality degradation.

G.729: This codec provides high-quality sound with a low bandwidth requirement. It may require as little as 12.8 kbps for a single line. The G.729 codec also offers a higher compression rate.

Opus: This codec is more adaptable to numerous applications. It supports bitrates between 6-510 kbps and 8-48 kHz sample rates.

Codec Selection & Compatibility

How do I select the right codec for my business?

No codec is perfect for all enterprises in all circumstances. In choosing the correct VoIP codec for your enterprise, you must consider several factors, including the following:

  1. Compatibility: Hardware compatibility is necessary because you must choose a codec that your VoIP hardware supports. Incompatibility will mean renting or buying additional hardware to support the codec
  2. Usability and cost: Some codecs are free and open-source, while others require a license. To use proprietary codecs, subscribe to a plan and pay the applicable licensing fees. Some proprietary codecs attract a higher cost for supporting hardware and fees charged based on call volume
  3. Bandwidth consumption: Some options better balance bandwidth usage and call quality. A good codec will consume just the bandwidth needed to maintain sound quality and not more
  4. Compression rate: A VoIP codec with a high compression rate delivers improved call quality. A codec that uses lossy compression techniques will eliminate unnecessary call data without degrading the sound quality

Go for a codec that works best for your needs while being a good fit for your budget. Factor in your network infrastructure and bandwidth capabilities. Also critical to consider is your enterprise’s average concurrent call volume.

The VoIP devices in your enterprise need the same codec to communicate with each other. When the transmitter and receiver channels in a VoIP call use different codecs that are not compatible, a gap is created. Transcoding, that is, the process of converting call media into a compatible codec for a VoIP device, bridges this gap. Transcoding is not only costly, but it can also cause latency. For this reason, ensure compatibility and uniformity when choosing a VoIP code for your enterprise.

Adaptive Codecs & Quality of Service (QoS)

Adaptive codecs can dynamically adjust to varying network conditions in real-time. This adaptability makes the codecs versatile and more suited for different communication channels. It optimizes the balance between bandwidth usage and audio quality. An excellent example of these adaptive, ultra-wideband codecs is Opus.

Quality of service is a set of techniques implemented to manage network traffic to ensure voice video and other critical applications receive priority over non-critical applications. Setting up network devices like switches and routers to prioritize traffic by various rules can prevent congestion.

Through traffic classification, shaping, and policing mechanisms, QoS ensures that voice traffic receives priority. It reduces latency issues, ensuring voice traffic is delivered with the necessary quality and reliability for a smoother communication experience while on call.

HD Voice & Enhanced Codecs

Enhanced and HD voice codecs, also known as wideband audio codecs, are emerging technologies in the VoIP codec market. The high-definition audio technology decreases background noise, delivering crystal-clear sounds. This superior audio quality provides a more immersive and natural communication experience than face-to-face interaction.

Impact on Business Communication

VoIP codecs help to deliver high-quality and reliable voice calls. With effective communication, you can look forward to enhanced collaboration among your team members and with relevant parties, such as suppliers, contributing to an improvement in overall productivity. Call audio quality will also influence customer interactions, and effective communication is one way to achieve customer satisfaction.

As mentioned, there is no perfect VoIP code for businesses. Compare what each offers against your business needs and how suited it is to your network infrastructure, and you can make an informed decision before buying a VoIP codec.


VoIP codecs play the two-fold role of improving call audio quality while conserving bandwidth. VoIP codecs enhance audio quality by reducing jitters, network latency, and data packet losses. That said, VoIP codecs are not created the same. For the best outcome, carefully consider your network conditions, specific communication needs, and budget when selecting a VoIP codec.

The chosen codec configuration will also optimize its performance to enjoy excellent call quality even under changing network conditions. Once set up, continuously monitor and test the call audio quality. Update firmware and software to current versions to resolve issues with codecs. Also, use QoS configurations of your routers or switches to prioritize VoIP traffic and minimize the jitter, latency, and packet loss that would otherwise degrade audio quality.

Hopefully, these highlights have helped you understand the role of VoIP codecs in in-call audio quality and how to select, configure, and optimize codecs for your VoIP solution. If you want to read reviews about VoIP providers and learn more about them, check out our blog and comparison pages for more insights.

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