Choosing the Right Codec and Quality of Service (QoS) for High-Quality Voice and Video Communications
March 7, 2023
In the realm of digital communications ensuring high-quality voice and video calls is crucial. However issues such as low bandwidth and Quality of Service (QoS) can often degrade call quality. To address these challenges several potential solutions are available in the realm of internet and technology. These include increasing available bandwidth, implementing proper QoS controls on edge equipment and firewalls to prioritize voice traffic, and utilizing codecs optimized for low bandwidth usage, such as G.729 or the adaptive Opus codec. By adopting these solutions, call quality issues can be mitigated, and a superior communication experience can be maintained.
Quality of Service (QoS) entails a set of techniques employed to manage network traffic effectively ensuring that critical applications like voice and video receive priority over non-critical applications. The objective of QoS is to reduce network congestion to guarantee the delivery of important traffic with the required level of reliability and quality. Implementing QoS involves configuring network devices such as routers and switches to prioritize traffic based on predefined rules. This is typically achieved through traffic classification, traffic shaping, and traffic policing mechanisms. Traffic classification involves categorizing packets as critical or non-critical based on their source, destination addresses, protocols, or application types. Traffic shaping involves controlling the flow of traffic to prevent congestion, while traffic policing enforces bandwidth limits and prevents non-critical traffic from overwhelming the network. By implementing QoS, network administrators can ensure that voice and video traffic receive priority treatment, leading to enhanced user experiences and reduced call quality issues.
The following adaptive codecs dynamically adjust the bitrate of encoded audio signals based on network bandwidth and other factors, optimizing the balance between audio quality and bandwidth usage. Adaptive codecs are particularly beneficial for real-time communication applications, where network conditions often vary, and bandwidth availability may be limited.
• AMR (Adaptive Multi-Rate): This codec which is designed for mobile networks provides high quality speech encoding at low bitrates. Operating at variable bitrates from 4.75-12.2 kbps it is widely used in 2G-4G mobile networks and is favored by major mobile network operators like AT&T and Verizon.
• AMR-WB (Adaptive Multi-Rate Wideband): Serving as a wideband extension of the AMR codec AMR-WB enables higher quality speech encoding at higher bitrates. It operates within a range of 6.6-23.85 kbps and finds common use in 3G and 4G mobile networks as well as VoIP and video conferencing applications. This codec is embraced by prominent mobile network operators and telecom vendors including Ericsson and Huawei.
• Opus Interactive Audio Codec: Engineered for real time communication applications Opus is a high quality, low latency codec for speech and audio encoding. It adjusts its bitrate from 6-510 kbps encompassing features such as voice activity detection, noise reduction and echo cancellation. Developed by Xiph.Org Foundation, Opus is widely used by cloud companies and telecom vendors, such as Discord and Riot.
The following codecs are tailored to deliver good voice quality while minimizing bandwidth usage making them suitable for scenarios with limited network bandwidth or low data rates such as dial-up connections or low bandwidth mobile networks.
• G.723.1: Operating at a fixed bitrate of 6.3 or 5.3 kbps this low bitrate codec provides moderate quality speech encoding which is commonly employed in VoIP and video conferencing applications and finds favor among cloud companies and telecom vendors including Cisco, Amazon Web Services, and Microsoft Azure.
• G.729: Another low bitrate codec delivering moderate quality speech encoding G.729 operates at a fixed bitrate of 8 kbps. It offers four different versions with varying features and complexities. This codec is utilized by cloud companies and telecom vendors such as Microsoft Teams, Zoom, and Cisco.
• iLBC (internet Low Bitrate Codec): Developed for VoIP and video conferencing applications, iLBC provides high-quality speech encoding at low bitrates. With fixed bitrates of 13.33 kbps or 15.2 kbps, it is commonly employed in low-bandwidth network environments. iLBC is employed by cloud companies and telecom vendors like Google Meet and Cisco, thanks to its creation by Global IP Solutions.
• SILK: Recognized for its low-latency and high-quality audio encoding SILK is a codec designed for real time communication applications. It offers high-quality speech and audio encoding leveraging variable bitrate (VBR) encoding within a range of 6-24 kbps, with a default bitrate of 12 kbps. SILK was developed by Skype and is widely utilized by cloud companies and telecom vendors, including Microsoft Teams, Zoom, and Google Meet.
Here’s a breakdown of other commonly used speech and audio codecs in the realm of digital communications:
• G.711: A low-complexity codec that provides high quality speech encoding, it operates at a fixed bitrate of 64 kbps and is commonly used in VoIP and PSTN systems. Cloud companies and telecom vendors such as Cisco, Amazon Web Services and Microsoft Azure extensively rely on this codec.
• G.722: This high-quality wideband codec ensures high-fidelity audio for voice and music, operating at a fixed bitrate of 48, 56, or 64 kbps it is widely employed in video conferencing, broadcast and audio recording applications. Prominent cloud companies and telecom vendors including Zoom, Google Meet, and Cisco opt for G.722.
• G.722.1: Offering high quality audio at lower bitrates G.722.1 is a low complexity wideband codec with a fixed bitrate of 24 or 32 kbps which finds common use in VoIP and video conferencing applications. Cloud companies and telecom vendors such as Skype, Cisco, and Google utilize G.722.1 extensively.
• G.728: Operating at a fixed bitrate of 16 kbps, G.728 is a low-bitrate codec providing moderate-quality speech encoding, it is widely employed in VoIP and video conferencing applications and favored by cloud companies and telecom vendors like Zoom, Cisco, and Google.
• Opus: A highly versatile codec known for its high quality speech and audio encoding Opus serves a wide range of applications. Supporting both constant bitrate (CBR) and variable bitrate (VBR) encoding it operates at bitrates ranging from 6-510 kbps. The choice of bitrate allows for a trade off between bandwidth usage and audio quality. Opus was developed by the IETF and is extensively utilized by cloud companies and telecom vendors including Microsoft Teams, Zoom, and Google Meet.
• AAC (Advanced Audio Coding): Designed specifically for high quality audio encoding AAC offers superior compression compared to MP3. It operates at variable bitrates ranging from 8-320 kbps and is commonly employed in digital music and audio streaming applications. Major streaming services and audio equipment manufacturers such as Apple Music, Spotify, and Sonos heavily rely on AAC.
• MP3 (MPEG-1 Audio Layer III): A widely used audio compression codec that provides moderate quality audio at low bitrates. It operates at variable bitrates ranging from 8-320 kbps and is commonly utilized in digital music and audio streaming applications. MP3 was developed by MPEG and is widely employed by streaming services and audio equipment manufacturers.
• Speex: A codec specifically designed for VoIP and video conferencing applications, Speex delivers high-quality speech encoding at low bitrates. It operates at variable bitrates ranging from 2.15 kbps to 44.2 kbps and incorporates features like variable bitrate encoding, voice activity detection, and noise reduction. Speex, developed by Jean-Marc Valin, is employed by open-source VoIP applications and telecom vendors.
It’s important to note that these are just a few examples of speech and audio codecs available and the choice of codec depends on specific application requirements, available network bandwidth and other factors, additionally many cloud companies and telecom vendors develop proprietary codecs or customize open source codecs to meet their specific needs.
Should your endeavors in the realm of digital communications require guidance in the pursuit of optimizing VoIP systems and enhancing audio quality, we extend an open invitation to connect with us. Discover how our expertise can assist you in unlocking the full potential of your communications systems by harnessing the power of bandwidth management to elevate your experience. Additionally we encourage you to explore our Hosted VoIP Services which provide a comprehensive solution tailored to meet your unique needs. Together let us embark upon a journey of transformation where the convergence of knowledge and innovation unfolds, paving the way towards a future defined by seamless and immersive communication.
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