VoIP Codecs Affect Call Quality
Have you ever wondered why sound quality in VoIP varies so much? Even when there's a good connection, sound quality can vary greatly with different VoIP codecs. This is because audio signals are compressed to reduce the consumption of network bandwidth required to transmit speech to the other party.
The compression and decompression are handled by special algorithms we call codecs (Coder-DECer). Some codecs provide better quality than others but may require more bandwidth. Telnyx strives to provide superior quality calling by supporting codecs that are optimized to maintain quality when there is lower bandwidth availability.
Let's cover the HD codecs we support here at Telnyx and their specifics.
G.711 – Is a narrowband audio codec that provides toll-quality audio at 64 kbit/s and is often used in telephony. It passes audio signals in the range of 300-3400 Hz and samples them at a rate of 8,000 samples per second. Its audio is considered to be high quality. There are two slightly different versions, μ-law, which is used primarily in the United States and Japan, and A-law which is in use in most other countries. G.711 can also be used for fax communication over IP networks.
G.729 – Compresses digital voice in packets of 10 milliseconds duration. Because of its low bandwidth requirements, G.729 is mostly used in VoIP applications where bandwidth must be conserved, such as conference calls. Standard G.729 operates at a bit rate of 8 kbit/s, but there are extensions, which provide rates of 6.4 kbit/s and 11.8 kbit/s for worse and better speech quality, respectively.
GSM – The true name of this VoIP codec is Full Rate but is most often referred to as GSM. The bit rate of the codec is 13 kbit/s, and the quality of the coded speech is quite poor by modern standards. However, at the time of development (early 1990s), it was a good compromise between computational complexity and quality. Gradually, Full Rate will be replaced by Enhanced Full Rate(EFR) and Adaptive Multi-Rate(AMR) standards, which provide much higher speech quality with lower bit rates.
Opus – Opus supports constant and variable bitrate encoding from 6 kbit/s to 510 kbit/s and five sampling rates from 8 kHz(with 4 kHz bandwidth) to 48 kHz(with 20 kHz bandwidth, the human hearing range). Opus allows the highest quality while maintaining low delay(has a very short latency of 26.5ms by default), which makes it suitable for telephony, VoIP etc.
For most carriers, the quality of voice will depend on the available bandwidth. If the amount of bandwidth available is low, then your typical provider will use VoIP codecs with a lower bandwidth consumption, resulting in a lower call quality.
At Telnyx, we do not limit calls on bandwidth, and always use the highest quality codec available for the call in order to provide our customers with the best calling experience.
Share on Social