ASUS (RT, GT, and Merlin firmware) Configuration Guide

Introduction

This guide provides our recommended configuration for ASUS routers (RT series, GT series, and Merlin firmware) to ensure optimal VoIP performance.

✅ Quick Checklist: Signs Your ASUS Router Needs VoIP Optimization

If you notice any of the following, this guide is for you :

Symptom	Description
One-way audio	You can hear the caller, but they can't hear you (or vice versa)
Phone disconnects after 15 minutes	VoIP desk phone loses registration after ~15 minutes
Registration failures	Phones won't register or stay registered
Dropped calls	Calls drop after specific time periods

✅ Compatible ASUS Models

Series	Popular Models	Notes
RT Series	RT-AC87U, RT-AC68U, RT-AC56U, RT-AC66U, RT-AC86U, RT-AX88U, RT-AX92U, RT-AX3000, RT-AX55, RT-AX52	Most popular, Merlin firmware support
GT Series (ROG)	GT-AXE16000, GT-AC5300, GT-AX11000	Gaming series, same SIP ALG location
N Series (Older)	RT-N56U, RT-N66U, RT-N15	Some models have no SIP ALG

Note: Some ASUS firmware versions (like 3.0.0.4.380.7378 on RT-N66U) have a bug where disabling SIP ALG stops all SIP traffic. Check your firmware version .

Provider-Specific Requirements

Important: You need to configure the following settings on your ASUS router.

Requirement	Specification
IP Range to Whitelist	`64.26.133.64/27`
Ports to Open	UDP `5060-5061` (SIP) and UDP `10000-20000` (RTP)
SIP Passthrough	Must be DISABLED

Core Configuration Steps

1. Disable SIP Passthrough (CRITICAL)

Critical: SIP Passthrough (SIP ALG) is enabled by default on ASUS routers and MUST be disabled for proper VoIP operation .

Web Interface Method	Log in to router	`http://192.168.1.1` or `http://192.168.50.1`
	Go to Advanced Settings	Click WAN in the left menu
	Click NAT Passthrough	Tab at the top of the WAN page
	Disable SIP Passthrough	Change SIP Passthrough to Disable
	Apply settings	Click Apply and reboot router
SSH/Telnet Method	Connect via SSH	Use a terminal client (PuTTY, Terminal)
	Check current value	`nvram get nf_sip` (should return "1")
	Disable SIP ALG	`nvram set nf_sip=0` then `nvram commit`

2. Port Forwarding for VoIP

Port forwarding must be enabled in the WAN configuration. The port range used by your VoIP phone must be assigned to its local IP address .

Navigation	Setting	Value
`WAN > Virtual Server / Port Forwarding`	Service Name	VoIP-SIP
	Protocol	UDP
	External Port	`5060-5061` and `10000-20000`
	Internal Port	Same as external
	Internal IP	Your VoIP device's IP address

3. IPTV/VoIP Configuration

ASUS routers have a dedicated IPTV section for VoIP configuration .

Navigation	Setting	Action
`Advanced Settings > IPTV`	Configure VoIP	Select ISP profile or manual settings
`Advanced Settings > IPTV`	UDP Proxy	Configure if required by your ISP

4. QoS for VoIP Priority

ASUS routers have built-in QoS to prioritize VoIP traffic .

Navigation	Setting	Action
`Adaptive QoS > Bandwidth Management`	Set VoIP priority	Click VOIP/Video Streaming to set highest priority
`Adaptive QoS > Bandwidth Management`	Apply settings	Click Finish and Apply

5. Important Model Notes

Model	Important Note
RT-N56U	No firewall masking SIP packets; works without interference. No SIP ALG to disable .
RT-N66U	Firmware 3.0.0.4.380.7378 has bug where disabling SIP ALG stops all SIP traffic. With latest firmware, leave SIP ALG enabled .
GT-AXE16000	Some users find no need to disable SIP ALG on newer routers; NAT Helper may be sufficient .

IP Whitelist Configuration

Create a port forwarding rule to allow traffic from our servers to reach your VoIP device.

Field	Value
Service Name	`VoIP-Provider`
Source IP	`64.26.133.64/27`
Destination IP	Your VoIP device's IP address
Port Range	`5060-5061` and `10000-20000`
Protocol	UDP

Note: The /27 notation includes 32 IP addresses ranging from 64.26.133.64 to 64.26.133.95.

✅ Configuration Checklist

#	Task
1	SIP Passthrough disabled in WAN > NAT Passthrough
2	Port forwarding configured for VoIP device IP
3	QoS enabled with VoIP priority
4	Whitelist rule created for `64.26.133.64/27`
5	Firmware checked for known bugs (RT-N66U specific)
6	Test call successful with two-way audio

Verification & Testing

SIP-ALG Verification Test

Test URL: http://mcstest.visualware.com/myspeed/myvoiph5_g711_sipalg.html

Item	Detail
Requirement	Download the BCS utility when prompted
Expected Result	`SIP ALG Firewall: N`

Functional Testing

Test Case	Procedure	Expected Result
Outbound Call	Place call to external number	Two-way audio established
Inbound Call	Call in from external number	Phone rings, call connects with two-way audio
Stability Test	Monitor for 15+ minutes	Phone stays registered, no disconnects

🚨 Troubleshooting

Issue	Likely Cause	Solution
One-way audio	SIP Passthrough still enabled	Disable in WAN > NAT Passthrough or via SSH
Phone disconnects after ~15 minutes	SIP ALG interference	Disable SIP Passthrough and verify with SSH
No SIP traffic after disabling ALG	Firmware bug (RT-N66U)	Update firmware or leave SIP ALG enabled
Port forwarding not working	IP address changed	Use DHCP reservation for VoIP device

🛠️ Official ASUS Tutorials

For visual guidance on configuring your ASUS router, visit the official ASUS support page:

🔗 ASUS Support