Evidently, over the last couple of decades, Voice over Internet Protocol (VoIP) services have revolutionized how we connect with others. And what lies at the heart of this transformation are VoIP codecs, which play a pivotal role in determining and shaping the quality of calls carried out through these services.
So, What are these codecs? In what ways do they affect the quality of calls? Well, that’s what this article is about!
In this article, we will examine what VoIP codecs are & their types, and discuss a bit about VoIP and Audio Quality as well, and give a brief overview of codec organization.
Let’s dive in!
What are VoIP Codecs?
To give a simple gist about VoIP codecs, you can think of these as a mechanism that compresses and transmits audio signals between devices at the same or different stations, thus enabling the transmission of simple, seamless, rapid, and clear conversations.
Now, to the technical bits- VoIP codecs, often interpreted as an acronym for Coder Decoder or Compression and Decompression, are computer programs that compress analog voice signals into digital data and then transmit them over the network.
Just another thing to note about VoIP codecs is that these algorithms are responsible for encoding and decoding audio signals during VoIP calls and determine how audio data is compressed for transmission and later decompressed for playback. They play a crucial role in Voice over Internet Protocol (VoIP) technology, ensuring that voice calls over the Internet maintain high quality while minimizing bandwidth usage.
VoIP and Audio Quality
If you, in one way or another, are involved in the VoIP world, you definitely should know how crucial it is to eliminate jitters when going through a call. Fundamentally, that’s the doing of the ‘codecs’!
Thus, logically speaking, a good VoIP codec will be able to compress audio data efficiently without sacrificing too much quality. This will ensure that your VoIP calls sound clear and natural, even if you are using a low-bandwidth connection.
However, you should note that other factors, like the network’s bandwidth and the quality of the hardware being used, also play a critical role in directly determining the quality of the call carried out via a virtual phone system. So be reasonable, and don’t try to solve every issue by playing with VoIP codecs.
Also learn: What Is WiFi Calling and How Does It Work?
Now, let us get down to the serious stuff!
Why are VoIP Codecs Important?
Reiterating what was said earlier, VoIP codecs are one of the few crucial factors that determine the quality of a call. And in the world of Cloud-based virtual phone systems, where everything depends upon real-time digital data transmissions, one of them being high-quality voice communications, the reduction of distractions like latency, jitter, background noise, and packet loss is a must.
Hence, that’s why VoIP codecs are important, as they directly impact the end user’s experience by eliminating the aforementioned factors. At the same time, it contributes to maintaining good VoIP QoS delivery irrespective of those external factors and ensuring a smooth and uninterrupted conversation.
How VoIP Codecs Improve Call Quality?
It is a widely accepted reality that VoIP calls regularly face issues like network fluctuations and varying internet speeds. To counter those issues, VoIP codecs are crucial to overcoming these challenges and maintaining call quality irrespective of those external factors. Here are a handful of examples of how these mechanisms improve call quality:
- A VoIP codec can remove background noise, like static and hum, from audio data. Thus making your calls sound clearer and more natural.
- Another one is echo cancellation, which is one of the most common VoIP problems. Guess what VoIP codecs can easily reduce the number of echoes.
- Lastly, by reducing jitter, VoIP codecs can help reduce audio crackle and distortion caused by variations in audio packet delays.
Well, after learning all that, one might ask, why is the importance of VoIP codecs often overlooked? Whatever the case, let’s move on to the types.
Types of VoIP codecs
There are nearly a dozen codecs that are used in the VoIP infrastructure, with each having its own way of working, characteristics, and application. But for now, we will discuss the most common and widely used one. So here are those:
G.711 is considered the gold standard codec known for its uncompressed audio transmission. While it offers high audio quality, it consumes more bandwidth compared to other codecs. It is offered and supported by almost all VoIP providers; an important reason for that is its high bitrate, which results in good audio quality.
With no noticeable latency, G.722 HD or High-definition codec can help to make your calls sound better by transmitting wideband audio signals. In simple words, this is a wideband codec that provides high-quality audio for HD VoIP calls by striking a balance between quality and bandwidth efficiency.
Next, we have G.729, a low-bandwidth codec that is often used for mobile VoIP calls. It was intended to serve those who just want decent audio quality with minimum bandwidth requirements. This codec is excellent at saving bandwidth, and although it sacrifices some audio quality, it ensures smooth communication no matter what the bandwidth is.
Lastly, let’s take a look at OPUS. It is a newer, versatile, and ultra-wideband codec that is designed to provide high-quality audio with low bandwidth usage with adaptive bit rates. What sets it apart is its ability to dynamically adjust to network conditions, making it suitable for various communication scenarios. One more thing: Opus is an open-source codec that requires no licensing fees for usage.
Organizing your codecs
And finally, let’s discuss about the configuration of codecs for network optimization. While, the majority of VoIP phones and softphones allow you to select the codec you want to use for your calls and let you configure it. Thinking of it as a piece of cake, right?
Here’s the problem: it involves configuring codecs for optimal network performance, adjusting settings based on network conditions, and ensuring that the chosen codec adapts to varying situations, maintaining call quality.
Frankly speaking, you don’t even have to do it yourself, as voice-over IP phone services do it for you by determining which codecs are the perfect ones for your available hardware. So, it is advisable to make the most of what is pre-prepared to the fullest.
Let’s recap: in this blog, we discussed what voice-over IP codecs are and their types, revealed the relationship between VoIP and audio quality, and discussed why codecs are important and how they can improve call quality.
By choosing the right codec for your needs, you can enhance the quality of your VoIP calls and make the most of this versatile technology.
As a closing thought, you should understand that these codecs are essential for ensuring that VoIP calls sound clear and natural and can be said to be the unsung heroes of clear and effective communication in this virtual telephony era.
Which codec is best for VoIP?
As VoIP codecs vary by provider, bandwidth requirements, and audio quality, there is no “best” codec. To be fair, you, yourself, do not need to select one specific codec for each call because most SIP and VoIP software providers support multiple codecs and automatically select the best one for each call.
However, if you are determined to choose one, here’s a simple comparison of the common ones to help you do so:
|Widely supported, high audio quality
|High bandwidth requirements
|Wideband codec for HD VoIP calls
|Not as widely supported as G.711
|Low-bandwidth codec, good for mobile VoIP calls
|Lower audio quality than G.711 or G.722 HD
|Newer codec, designed for high-quality audio with low bandwidth usage
|Not as widely supported as G.711 or G.729
What is the most common SIP codec?
G.711 is the most common SIP codec. This codec is commonly used because it is the same codec used by the PSTN on top of that, it is able to provide high-quality audio at relatively low-bandwidth, making it appropriate for most VoIP applications as well. It is an open-source standard codec that is widely supported by VoIP devices and softphones.
What is the audio codec in VoIP?
Audio codecs in VoIP are used to determine the quality of audio, bandwidth, and compression & decompression of VoIP phone calls. With VoIP codecs, audio signals can be compressed into data for transmission over the internet and then decompressed back into uncompressed audio signals using codecs.
Is G722 better than G711?
When comparing both, G.722 is generally believed to be better than G.711 in terms of audio quality. That is because, as a wideband codec, G.722 offers a wider frequency range (50Hz to 7kHz) than G.711(200 Hz to 3.4 kHz), allowing for a more realistic sound by capturing and reproducing more of the finer details of human speech.
Furthermore, G.722 has a lower latency than G.711, which is a crucial factor for VoIP calls because it can reduce the feeling of being disconnected or “on the phone.”