From my experience, 'configure QoS for voice' is advice I see in every VoIP article and almost nobody explains how. It is not a tick-box. It is a multi-layer problem. Let me walk through what actually works on a typical UK SMB network.
The 4 layers where QoS needs to happen
| Layer | What it does | Where it breaks |
|---|---|---|
| Endpoint (phone / softphone) | Tags voice packets with DSCP EF | Default config on most IP phones is 'best effort' — must configure |
| LAN switch | Recognises DSCP EF, puts in priority queue | Most unmanaged switches ignore DSCP entirely |
| WAN router | Priority queue outbound, rate-limits bulk traffic | Consumer routers either do not support this or do it badly |
| ISP upstream | Honours DSCP end to end | Most UK retail ISPs strip DSCP markings at the handoff |
That last one is the bitter truth nobody mentions. BT, Virgin Business, TalkTalk Business, Sky Business — all of them strip DSCP on the consumer-grade products. You need a business-grade product (BT Business fibre with QoS, or dedicated SIP trunks from a tier-1 provider) for end-to-end marking to survive.
The minimum that works for most UK SMBs
Assuming a typical 25-person office with a MikroTik or UniFi router, 100 Mbps symmetric fibre, and mixed IP phones + softphones. Honestly this is the config we recommend most often.
Step 1: Endpoint DSCP marking
All major IP phone brands support this. Examples.
Yealink T4x / T5x series: Web UI → Features → Voice → SIP DSCP = 46 (EF), RTP DSCP = 46 (EF). Save + reboot phone.
Polycom VVX: Web UI → Network → QoS → SIP DSCP = 46, RTP DSCP = 46. Save.
Cisco SPA / MPP: Voice → System → Network Settings → Signal DSCP = 46, Media DSCP = 46.
Snom: Setup → Advanced → QoS → Signal = 26 (AF31), Media = 46 (EF).
Softphones: Most UK VoIP provider softphones (Bria, Zoiper, provider branded) mark correctly by default. Check Settings → Advanced → DSCP. If it says 0, that is wrong, set 46 for media.
Step 2: LAN switch priority queue
Unmanaged switches ignore DSCP. If you have a managed switch (Netgear GS308EP, TP-Link TL-SG108E, UniFi USW-Lite-8-PoE, Cisco SG350), enable QoS with DSCP trust mode. Example config on a UniFi switch, UI path Settings → Switches → the switch → QoS → Enable 802.1p / DSCP.
On unmanaged switches, you physically cannot do this. Either replace the switch or accept that under heavy load, voice and data will compete.
Step 3: WAN router priority queue
This is the one that actually matters on a congested UK broadband line. Two approaches.
Approach A, priority queue by DSCP (preferred):
On MikroTik:
/queue tree add name=voice parent=global packet-mark=voice priority=1 max-limit=10M
/ip firewall mangle add chain=forward dscp=46 action=mark-packet new-packet-mark=voice
On UniFi Dream Machine, UI path Settings → Internet → Smart Queues → enable WAN rate + priority voice traffic.
On Draytek 2862/2926, UI path Bandwidth Management → Quality of Service → set Voice profile 46 to High.
Approach B, rate-limit bulk traffic (if you cannot do A):
Reserve 80% of upload for VoIP, 20% for bulk. Brutal but works. Each call uses about 100 kbps, so 80% of 50 Mbps upload = 40 Mbps = 400 concurrent calls. Plenty for any SMB. This is the fallback for cheap routers that do not support DSCP.
Step 4: ISP upstream (the honest part)
With retail UK broadband, your QoS markings stop at your router. Beyond that, the public internet treats your packets as best-effort regardless of DSCP. This is fine 99% of the time because backbone networks are rarely congested. It becomes a problem when:
- Your ISP has local congestion (shared cable, old DSL)
- Your provider's SIP trunk goes through a peering point that is saturated at peak
- You are over asymmetric fibre and upload is maxed by a backup
The fix is to buy a business-grade internet product that includes a CoS contract, or use SIP trunks from a provider that has direct peering with your ISP. For example, some UK VoIP providers peer directly with BT, Virgin Business, and Sky to avoid the public internet entirely between your office and the PBX.
Testing your QoS actually works
Run the MTR test for 60 seconds while saturating your upload with a large file transfer.
mtr --report --report-cycles=60 --tcp --port=5060 <your SIP provider IP>
Look for packet loss on the MTR output. If loss is < 0.5% and latency variance (jitter in column) is < 10ms, QoS is working. If you see > 2% loss or jitter > 30ms, voice will sound bad under load.
What a good VoIP provider should do for you
Honestly, not gonna lie, the whole QoS stack is painful to configure. A decent provider will ship you a pre-configured router with QoS already set up for their SIP trunks. We do this at DialPhone for customers on our managed hardware option, about £8 per month per router. If your current provider just says 'enable QoS on your router' and leaves you to it, that is not support, that is homework.
The single best thing you can do, if nothing else, is upgrade from consumer BT Hub to a MikroTik hAP ax². That one change fixes about 70% of UK small-business VoIP quality issues before you even think about DSCP.
