How the code works
During every VoIP call, the codec on the caller's side captures the voice, encodes it into data packets and sends them over the network. On the recipient's side, another codec receives the packets and decodes them back into sound. The entire process takes place in real time, encoding must not add unnecessary delay.
Codecs differ in sampling frequency, compression algorithm and resistance to packet loss. A higher sampling frequency transmits the full range of the human voice more faithfully, but at the cost of greater data demand. Compression algorithms reduce this demand, sometimes at the expense of the resulting audio quality.
Overview of codecs G.711, G.722 and Opus
- G.711 - classic standard originally designed for fixed telephone networks. It does not use compression, so it transmits voice faithfully but consumes approximately 80-100 kbps per call. It is supported by virtually all SIP devices and operators in the Czech Republic.
- G.722 - Wideband codec with a sampling frequency of 16 kHz compared to 8 kHz for G.711. The voice sounds more natural, and lower frequencies are not cut off. Data demand is comparable to G.711.
- Opus - modern open codec with variable transmission rate. It dynamically adapts to network quality, handles packet loss and performs well even with unstable connections. It is now the standard in WebRTC communication.
How to choose a codec for business telephony
For companies with a sufficiently robust fixed connection, G.722 is the natural choice. It offers better voice quality than G.711 at comparable data demand. G.711 is suitable where you need maximum compatibility with older equipment or external partners.
Opus is particularly valued by companies whose employees make calls via mobile data or from remote sites with fluctuating network conditions. In the event of partial packet loss, it maintains call intelligibility better than older codecs.
When calling the public telephone network, the codec agreed with the SIP operator is decisive, cloud PBXs usually handle this agreement automatically. On virtualni-ustredna.cz, the preferred codec can be set in the administration without needing to modify the device configuration.
Codes and conference calls
During conference calls, all participants must use the same codec. If one device does not support G.722, the switchboard switches the entire call to G.711. The result is lower quality, even though other phones can handle G.722.
If you find that conference call quality is worse than a direct call, check the codec settings in your SIP client or IP phone. An improperly configured codec is one of the most common causes of quality issues not caused by the network.
Frequently Asked Questions
Which codec provides the best voice quality?
For standard business telephony, G.722 is a good choice due to its more faithful voice transmission. Opus offers an advantage in unstable networks. The best codec is the one supported by both parties of the call and matching your connection speed.
How much data does a single VoIP call consume?
For G.711, approximately 80-100 kbps per call including protocol headers. G.722 is similar. With a good network connection, Opus achieves comparable quality at a significantly lower data rate, typically around 20-40 kbps.
Can different codecs be used in a single call?
No: both ends of the call must operate with the same codec. Therefore, when establishing a call, the PBX or gateway performs negotiation (SDP handshake) and selects a codec supported by both parties. If no agreement is found, the call will not be established.