Beginners Guide to VoIP Audio CODECs

Posted on

VoIP Network codec symbols

A Voice over Internet Protocol (VoIP) network enables the transmission of voice calls through the Internet or other digital networks. These audio signals have to be encoded into digital data to be transferred over an IP network and vice versa.

To make the transmission of audio signals faster, smoother, and improve the overall calling experience, VoIP systems have to compress digital data. The device that helps these systems perform this function is known as CODEC (encoder-decoder).

Although you will find a range of CODECs for different communication systems – audio, text, video or fax – certain CODECs are used in VoIP networks. In this article, we have highlighted the main types of audio CODECs for the VoIP environment.

G.711

Introduced in 1972 by ITU, the G.711 CODEC has been used in digital telephony since then. The Codec has two main variants:

  • A-Law (used in Europe and in international telephone links)
  • U-Law (used in the U.S and Japan)

The G.711 CODEC uses a logarithmic compression to encode and decode audio data. It compresses every 16-bit sample to 8 bits and therefore has a compression ratio of 1:2. This means that the bit rate for one direction is 64 kbit/s, while a two-way call consumes 128 kbit/s in total.

Compared to other CODECs, G.711 offers better voice quality. Some experts suggest using this CODEC because it offers the same voice quality as a digital telephone.

Since the G.711 CODEC has no licensing fee, you can apply it to any VoIP application without incurring additional costs. It works best in a local area network as these networks have additional bandwidth available.

Moreover, the implementation of the CODEC is straightforward. You can implement the CODEC by a simple table lookup, so it doesn’t require much CPU power. Due to its exceptional call quality, this CODEC has a Mean Opinion Score (MOS) of 4.2. However, it doesn’t compress data as much as other CODECs; it requires higher network resources.

OPUS

OPUS is an ideal option if you want to transmit high-quality audio with the help of state of the art compression techniques. The codec is extremely flexible, and you can use it for both high fidelity audio and clear speech.

Although the codec was designed for WebRTC in the beginning, it’s now used outside the scope of browser-based telephony. The greatest example of this is OPUS’s acceptance from many VoIP services into their SIP telephones.

Furthermore, it allows you options such as wireless audio, voice recording, WebRTC. You can also leverage OPUS to take care of low latency issues. However, the codec still needs to address issues related to lossless audio compression.

G.729

While the G.729 CODEC needs a low bandwidth for operations, it still provides decent audio quality. After slicing audio data into frames, the CODEC encodes each frame separately. Usually, each audio frame has a length of 10 milliseconds. For a sampling frequency of 8 kHz, a 10 ms frame contains 80 audio samples.

Since the G.729 algorithm encodes each frame to 10 bytes, the resulting bit rate is 8 kbit/s for a single direction. When it is used in VoIP, about 3-6 frames are sent in every packet. This needs to be done because the overhead of packet headers (IP, UDP, etc.) is 40 bytes and administrators have to send as much“useful” information as possible.

However, G.729 CODEC isn’t free. You may have to pay for using this licensed CODEC. However, end users can get around the licensing fees by buying hardware that implements the CODEC (VoIP phone or gateway). Since the manufacturers of the hardware already pay the license fee, users buying those devices won’t have to.

Alternatively, administrators can also use a variant of G.729 known as G.729a. The variant is wire-compatible with the original G.729 CODEC, but it needs lower CPU requirements and is, thus, more efficient.

G.723.1

The G.723.1 CODEC was developed after ITU announced a competition to design a CODEC algorithm that allows calls over 28.8 and 33 kbit/s modem links. At the end of the competition, ITU comes up with two solutions, and it decided to use both of them as different variants.

Both variants of the G.723.1 operate on audio frames that are 30 milliseconds long. This means that both variants can process 240 samples. However, the algorithms for these variants are different.

The first variant has a 6.4 kbit/s bit rate, along with a reasonable MOS value of 3.9. The second variant, however, has a lower bit rate (5.6kbit/s) and MOS (3.7). The encoded frame of the first variant is 24 bytes long, whereas the length of the second variant is 20 bytes.

Although G.723.1 doesn’t have the best call quality, it has the best compression ratio (12:1) among all other CODECs. As a result, administrators using this CODEC can manage connections on a low-bandwidth network.

G.722

G.722 is an initial codec issued by ITU Telecommunication Standardization Sector standard in 1988. The codec operates bands including 48, 56 and 64 Kbps. It uses a codec technology that relies on a sub-band of ADPCM (Adaptive Differential Pulse Code Modulation).

The codec samples data at a rate of 16 kHz. While this may not seem much at first glance, it is actually twice as fast as a traditional telephony interface. As a result, G.722 manages to produce superior audio quality and clarity.

G.722 also represents the wideband audio coding system. This means that it can be used for various high-quality speech applications such as high-quality VoIP.

One of the main applications for G.722 in VoIP is that it facilitates local area networks that have network bandwidth easily available. In such cases, the codec offers greater improvement in speech quality compared to narrow-band codecs like G.711, without increasing the complexity in implementation.

GSM 06.10

Designed by the European Telecommunications Standards Institute, GSM 06.10 was originally meant for GSM mobile networks. But, its variant is also used in open source VoIP applications.

The CODEC operates on audio frames that are 20 milliseconds long (e.g., 160 samples). It compresses each frame by 33 bytes and has a bit rate of 13 kbit/s. However, 4 bits in each frame remain unused as the encoded frame is 32 and a half-byte long. GSM 06.10’s audio quality is reasonable, and the CODEC’s MOS is 3.7.

Conclusion

While each CODEC used in a VoIP environment has its pros and cons, some CODECs (like G.711) have better audio quality, whereas others prioritize the compression of data over others. Therefore, before selecting a CODEC, you must evaluate your specific requirements and goals to ensure you invest in one that can optimize your VoIP performance.

DLS Internet Services provides a comprehensive VoIP solution to present Unified Communications as-a-service (UCAAS). The company implements the best practices in VoIP and has been delivering innovative business-to-business PBX solutions since 2004.

Currently, DLS offers VoIP services such as hosted PBX and virtual PBX phone systems to help organizations optimize their communication. Moreover, the company is a unique business Internet Service Provider that allows it to ensure quality connections between VoIP peers. The Internet service, coupled with the company’s powerful VoIP system, offers the perfect communication toolkit for your organization.

Top