Verified against Ofcom Connected Nations Spring 2026, IASME Cyber Essentials April 2026 standard and current operator pricing.
Quick answer: SIP ALG is a router feature that tries to help VoIP traffic but almost always breaks it. If your VoIP phones randomly lose registration, calls have one-way audio, or you can’t receive inbound calls — there’s a 90% chance your router has SIP ALG enabled. Turn it off. Step-by-step instructions for BT Smart Hub 2, Draytek, Cisco, FortiGate, Ubiquiti, MikroTik, Netgear and Sky Q hubs are below.
What is SIP ALG, in plain English?
SIP ALG stands for Session Initiation Protocol Application Layer Gateway. It’s a router feature that:
- Watches outgoing VoIP packets,
- Looks at the IP addresses inside the SIP messages,
- Rewrites them so they appear to come from your public IP rather than your private LAN IP,
- And opens the necessary ports for the return audio.
The intention is good — help VoIP traffic survive NAT (Network Address Translation). The reality is that it was designed in the early 2000s for a SIP world that no longer exists, it’s poorly implemented in most consumer-grade routers, and modern VoIP providers handle NAT traversal far better themselves.
SIP PhoneHosted PBXPSTN / SIP Trunk
When SIP ALG mangles a packet, the audio path between phone and PBX is broken — even though signalling looks fine.
The symptoms — does this sound like your office?
- VoIP phones lose registration every 60 seconds and re-register, then drop again.
- You can dial out but nobody can call in — phones don’t ring on inbound calls.
- One-way audio — you can hear them, they can’t hear you, or vice versa.
- Calls drop after 30 seconds — exactly when SIP “session refresh” times out.
- Transferring calls fails — when you press transfer, the call disconnects entirely.
- DTMF tones (press 1 for sales) don’t reach the IVR — your menu choices are ignored.
- Inbound CLI shows wrong number — caller ID is mangled.
- “Connection lost” errors in your softphone every few minutes.
If you tick three or more of those, SIP ALG is almost certainly the culprit.

How to disable SIP ALG — by router brand
BT Smart Hub 2 (most BT Business Broadband customers)
- Browse to
http://192.168.1.254/and log in (default password is on the back of the hub). - Go to Advanced Settings → Firewall → SIP ALG.
- Uncheck “Enable SIP ALG”, click Save.
- Reboot the hub from the same admin page.
Draytek Vigor (2762, 2865, 2927)
- Log in to the Vigor’s web UI (default
192.168.1.1). - Go to NAT → ALG.
- Untick “Enable SIP ALG”, click OK.
- Reboot from System Maintenance → Reboot System.
Cisco Meraki MX (any model)
- In the Meraki dashboard, open Security & SD-WAN → Firewall.
- Scroll to SIP ALG, set to Disabled.
- Click Save — it takes effect within seconds, no reboot required.
FortiGate (60F, 100F, etc.)
- SSH or use the GUI to access the FortiGate.
- Run
config system settingsthenset sip-helper disableandset sip-nat-trace disable. - Run
config system session-helper, find the SIP entry (show), anddelete [number]. - Save with
end. Reboot recommended.
Ubiquiti UniFi / EdgeRouter
- SSH into the EdgeRouter / UniFi gateway.
- Run
configurethendelete system conntrack modules sip. - Commit and save:
committhensavethenexit.
MikroTik RouterOS
- In WinBox or the web UI, go to IP → Firewall → Service Ports.
- Find
sip, double-click, untick Enabled, click Apply.
Netgear (Nighthawk, Orbi)
- Log in at
http://routerlogin.net. - Go to Advanced → Setup → WAN Setup.
- Scroll to “Disable SIP ALG” and tick the checkbox.
- Apply — reboot if prompted.
Sky Q hub / Sky Broadband router
Sky disables SIP ALG by default on modern hubs. If you’re on an older hub, the only fix is to replace it with a third-party router (Draytek 2865 is the go-to for UK SMBs).
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How to verify SIP ALG is really off
1. Restart the VoIP phone
Power-cycle a desk phone or restart the softphone app. It re-registers fresh — if it stays registered for 5 minutes, you’re past the most common SIP ALG failure mode.
2. Make a 2-minute test call
SIP ALG often kicks in at 30 seconds (session refresh timer). A clean 2-minute call with audio both ways = working.
3. Test DTMF
Call a number with an IVR (try 0800 800 150 — BT’s automated line). If the menu responds when you press 1/2/3, DTMF is reaching the carrier.
4. Use sngrep
For technical teams: capture SIP packets with sngrep on a Linux box plugged into the LAN. Compare INVITE Contact headers — the IP should match your public WAN, not be mangled.
Why VoIP providers don’t need SIP ALG anymore
SIP ALG was designed for an era when every SIP endpoint had to manage its own NAT traversal. In 2026, modern hosted VoIP services use:
- STUN — endpoints discover their public IP themselves, so the SIP messages already contain the right addresses.
- ICE — endpoints negotiate the best media path automatically (works through almost any NAT).
- Symmetric RTP — providers send audio to whichever IP/port the audio came from, NAT just-works.
- Far-end NAT detection — the SBC at the carrier figures out NAT without help from your router.
Result: SIP ALG is at best redundant, at worst actively destructive. Every major UK VoIP provider (hypercloud, RingCentral, 8×8, Vonage Business, Microsoft Teams Phone direct routing) explicitly recommends turning it off.
VoIP still flaky after disabling SIP ALG?
Talk to our network team — we’ll check QoS, DSCP marking, codec choice and SBC config to find the real cause. Free 30-minute review.
SIP ALG — FAQs
Historical inertia. It was useful on early-2000s consumer routers with poor NAT implementations. Most modern firmware leaves it enabled to avoid breaking legacy customer setups, but for any 2026 hosted VoIP service it should be off.
Rarely. Some legacy on-premise PBXs without STUN support behind double-NAT might benefit. For 95%+ of UK businesses on hosted VoIP it just causes problems.
No — SIP ALG only inspects SIP packets (UDP/TCP 5060, 5061). Disabling it has zero impact on web browsing, email, video conferencing or any other traffic.
Check (1) QoS is marking SIP/RTP DSCP EF, (2) your firewall isn’t aggressive on UDP timeout (set ≥ 90 seconds for UDP), (3) you have enough upload bandwidth (100 kbps per call), (4) your provider’s SBC IP isn’t being filtered. Or ask us — these checks take 30 minutes.
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