Glossary · VoIP and call quality
Jitter
In one sentence
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.
Also known as : network jitter · packet jitter · VoIP jitter
What is jitter?
In a VoIP call your voice is sent as small packets at a steady pace, usually one every 20 milliseconds. Ideally they would arrive at the same steady pace. In reality some take longer than others: they queue behind a large download in a router, wait for the Wi-Fi to free up, or take a different path. That unevenness is jitter.
It is easy to confuse with latency. Latency is the average delay; jitter is how much the delay varies around that average. A call can have perfectly acceptable latency and still sound dreadful because of jitter.
How it works
To cope, every phone and softphone has a jitter buffer. It holds incoming packets for a few tens of milliseconds, puts them back in order and plays them out at a constant rate.
- A packet that arrives within the buffer window is played normally.
- A packet that arrives too late is thrown away, exactly as if it had been lost. You hear a gap, a clipped word or a metallic warble.
- Adaptive buffers grow when jitter rises, which smooths the audio at the cost of a little more delay.
Jitter is measured in milliseconds and reported in the RTCP statistics exchanged during the call and in the call-quality reports of most phone systems. It weighs heavily in the MOS score. The main fix is QoS: prioritising voice packets on the office network and the internet link so they never wait behind bulk traffic. Wired connections for desk phones help too, because Wi-Fi shares airtime and retransmits frames.
Example
A veterinary practice in Swansea runs a cloud phone system on the same broadband as its practice-management software, which uploads X-ray images to the cloud. Receptionists notice that calls break up whenever images are being sent. The call-quality reports show jitter jumping above 70 ms during uploads. Enabling voice prioritisation on the router and capping the upload rate of the imaging software brings jitter back under 10 ms, and clients stop asking staff to repeat themselves.
Jitter at Universal Telecom
Calls on our virtual PBX and SIP trunk can be reviewed using their quality statistics. If your calls sound choppy, our team can help you check the local network and prioritisation settings. Contact us for a diagnosis.
Frequently asked questions
What is an acceptable jitter level for VoIP?
Most providers aim for average jitter under 30 ms. A jitter buffer can absorb occasional spikes, but sustained high jitter turns into discarded packets and broken-up audio.
What is the difference between jitter and latency?
Latency is how long packets take to arrive on average; jitter is how much that time varies from packet to packet. A call can have acceptable latency and still sound choppy because of jitter.
Can a VPN cause jitter?
It can. Encrypting and routing voice through a VPN concentrator adds processing and an extra hop, and if that device is busy, packets queue unevenly. Keeping voice outside the tunnel often helps.
Related terms
- Latency
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.
- 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.
- 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.
- 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.
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