Glossary · VoIP and call quality
QoS (quality of service)
In one sentence
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.
Also known as : QoS · quality of service
What is QoS?
By default a data network treats every packet the same way: first come, first served. For a file download, half a second of delay goes unnoticed. For a VoIP call you hear it straight away, as robotic audio, words cut off or a lag between speakers.
Quality of service is the umbrella term for the mechanisms that let some traffic jump the queue. Carriers also use “QoS” for contractual commitments such as availability or fix times, but in a phone system project it usually means how your own network is configured to protect calls.
How it works
QoS works in three stages:
- Classify. The IP phone or softphone tags its packets with a DSCP value in the IP header, typically EF for audio. On larger networks, phones also sit on their own voice VLAN.
- Queue. Switches and the router put tagged packets in a priority queue that is served first.
- Shape. On the internet link, part of the upload bandwidth is reserved for voice, or other traffic is capped so there is always room left for calls.
These settings target the three faults that ruin calls: latency, jitter and packet loss. The result shows up in the MOS score that most phone systems and monitoring tools report.
The catch is that QoS only applies where you control the equipment. Once packets leave your router for the public internet, markings are usually ignored. The good news is that the most common bottleneck, the upload side of your broadband, is exactly where QoS helps most.
Example
A recruitment firm in Leeds with 30 staff on a single FTTP line noticed choppy calls every morning at nine. The cause was laptops syncing large files to the cloud as people logged in. Their IT partner moved desk phones and softphones onto a voice VLAN, set EF marking, and configured the firewall to give voice a guaranteed slice of upload bandwidth. Morning syncs now take slightly longer, but the call quality complaints stopped and average MOS rose back above 4.
QoS at Universal Telecom
Our voice services run over the internet connection you already have, so QoS is configured on your side; we share the recommended markings and ports during set-up. If you connect your own PBX through our SIP trunk, you can choose between common codecs to balance quality against bandwidth. Multi-site networks or strict firewalls may also call for a session border controller. For advice on your own network, get in touch.
Frequently asked questions
Does QoS work over the internet?
Not end to end. DSCP markings are honoured on your own network and sometimes on a private carrier circuit, but the public internet gives no priority guarantees. QoS still matters where congestion usually happens: the upload side of your own router.
Which DSCP value should voice use?
The common convention is EF (46) for the RTP audio stream and CS3 or AF31 for SIP signalling. What matters most is that phones, switches and the router all apply the same policy.
Do I need QoS on a fast fibre line?
More bandwidth lowers the risk but doesn't remove it. A cloud backup or a large upload can still fill the upstream link for a few seconds, and QoS stops those spikes from turning into gaps in the middle of a call.
Related terms
- 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.
- 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.
- 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.
- 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.
- 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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