What latency consists of
The total latency of a VoIP call is the sum of delays from all transmission phases:
- Encoding latency - time required for the codec to capture and encode a voice sample before transmission.
- Network latency - the physical distance between your device and the central hub's server, plus latency on individual routers.
- Latency of the jitter buffer - The buffer memory in the phone or software adds a short delay to compensate for network instability.
- Decoding latency - the time required for the receiver to decode the packets back into sound.
These partial values are summed to yield one-way call latency. A person perceives both sides - that is, the time there and back.
What latency is acceptable
The recommended one-way latency for VoIP calls is below 150 ms. This value is also stated in the international ITU-T G.114 recommendation. At a latency of 150-300 ms, the call still functions, but pauses are noticeable. Above 300 ms of one-way delay, the call becomes unnatural and communication is uncomfortable.
For comparison, a satellite call has a latency of around 500-600 ms and the echo is clearly audible. Data centres for cloud-based central units located in Europe enable a latency of under 50 ms with a fixed connection without any special measures.
How to reduce latency
Latency is primarily influenced by your choice of infrastructure and network configuration:
- Choose a cloud central station provider with a data centre in Europe, physical distance directly affects network latency.
- Prefer a wired Ethernet connection over Wi-Fi, which introduces its own latency and instability.
- Configure QoS on the router so that voice packets do not have to wait behind other traffic.
- Avoid unnecessary VPN gateways in the voice traffic path: each encryption layer adds latency.
On virtualni-ustredna.cz the servers are located in the Czech Republic and Germany, so latency for domestic companies is usually low even without special network measures.
Frequently Asked Questions
What latency is acceptable for VoIP?
One-way latency under 150 ms is optimal. Calls function without significant issues even with latency up to 300 ms, but pauses are noticeable. Above 300 ms one-way, communication becomes uncomfortable and participants interrupt each other.
How do I measure latency?
The basic measurement is performed using the ping command to your PBX server address. The resulting response time is approximately double the one-way latency. Accurate results for VoIP are provided by specialised tools or the built-in quality monitoring in a cloud-based PBX.
Does latency affect codec selection?
Yes. Every codec processes voice in short time intervals, and their processing adds so-called algorithmic latency. With modern codecs such as G.711 or G.722 this value is very small. Distance from the server and network status have a greater impact on total latency.