Glossary · VoIP and call quality
Latency
In one sentence
Latency is the time your voice takes to travel from your mouth to the other person's ear. In VoIP, once one-way delay passes about 150 ms, conversations start to feel stilted.
Also known as : network latency · call latency · one-way delay · end-to-end delay
What is latency?
Latency is the delay between the moment you say something and the moment the other person hears it. On old copper lines it was too small to notice. In VoIP, your voice passes through several devices and networks, and each one adds a few milliseconds.
High latency does not drop calls or make the sound crackly. What it does is wreck the rhythm of a conversation: both people start talking at once, stop, apologise and wait. It also makes any echo more noticeable, because it comes back later.
How it works
End-to-end latency is the sum of several delays:
- Encoding: the codec collects a frame of audio, often 20 ms, before sending it, plus processing time.
- Propagation: physical distance. London to Sydney adds well over 100 ms through fibre on its own.
- Network equipment: routers, firewalls, VPN concentrators, session border controllers and any transcoding step.
- The jitter buffer: the receiving phone holds packets for tens of milliseconds to put them back in order.
The reference point is ITU-T Recommendation G.114: up to roughly 150 ms one way, most users notice nothing; beyond 400 ms conversation becomes difficult. Latency also feeds into the MOS score used to rate call quality.
In practice, distance is rarely the culprit in a business. The usual causes are a saturated internet link (backups, cloud sync, software updates), congested Wi-Fi, or a VPN that routes voice via head office before it reaches the provider.
Example
An insurance broker in Belfast moves its staff to hybrid working. Customers complain that conversations feel awkward and full of interruptions. Measurements show about 260 ms of one-way delay: every softphone call is backhauled through the corporate VPN to the head-office firewall and out again. Excluding voice traffic from the tunnel (split tunnelling) lets softphones reach the provider directly, and latency drops below 80 ms. The complaints stop within a week.
Latency at Universal Telecom
Our virtual PBX and SIP trunk provide the call statistics you need to diagnose delay. If calls feel out of step, our technical support team can help you pinpoint whether the cause lies in the office network, a VPN or the internet link. Contact us with a few examples of affected calls.
Frequently asked questions
What is an acceptable latency for VoIP?
ITU-T Recommendation G.114 treats up to about 150 ms one way as comfortable for most users. Between 150 and 400 ms people start talking over each other; beyond 400 ms conversation becomes hard work.
Is latency the same as ping?
Not quite. Ping measures a network round trip between two machines. Call latency is one-way and also includes codec processing and the jitter buffer, so it is always higher than half the ping time.
Does a faster broadband package reduce latency?
Only if the line was saturated. Speed is capacity, not delay. Prioritising voice traffic and keeping it out of VPN tunnels usually does more than upgrading bandwidth.
Related terms
- Jitter
Jitter is the variation in the arrival time of voice packets. When it exceeds what the phone's jitter buffer can absorb, packets are discarded and the voice sounds choppy or robotic.
- MOS (mean opinion score)
MOS (mean opinion score) rates perceived call quality on a scale of 1 to 5. In VoIP it is estimated from latency, jitter, packet loss and the codec in use, without anyone listening.
- QoS (quality of service)
QoS (quality of service) is the set of network settings that give time-sensitive traffic, such as voice, priority over other data, so calls avoid delay, jitter and packet loss when the link is busy.
- Codec
A codec is the algorithm that digitises and compresses your voice during a VoIP call and turns it back into sound at the other end. It sets both the audio quality and the bandwidth used.
- VoIP
VoIP (voice over internet protocol) carries phone calls as packets of data over an IP network, such as the internet or an office LAN, instead of over the copper lines of the traditional phone network.
- SBC (session border controller)
A session border controller (SBC) sits at the edge of a network and controls SIP calls going in and out: it filters traffic, hides the internal network, handles NAT and can transcode or encrypt calls.
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