What is One-Way Audio?
One-way audio occurs when you can hear the other party but they cannot hear you, or vice versa. It is one of the most common VoIP issues and is usually caused by network configuration problems.
You hear them, they don't hear you
Your outbound RTP packets are blocked. Usually firewall or NAT issue on your side.
They hear you, you don't hear them
Inbound RTP packets are blocked. Check your firewall or NAT settings.
Common Causes
1. NAT (Network Address Translation)
Most common cause. Your router translates private IPs to public IPs, but RTP packets use different ports than the SIP signaling, confusing the router.
2. Firewall Blocking RTP
SIP uses port 5060, but voice (RTP) uses ports 10000-20000. If your firewall only opens 5060, calls connect but have no audio.
3. SIP ALG (Application Layer Gateway)
Many routers have SIP ALG enabled by default. It often corrupts SIP packets instead of helping. Disable it.
4. Incorrect NAT Settings in PBX
Your Asterisk/FreePBX needs to know its external IP to properly route RTP traffic.
Diagnosing the Problem
Quick Diagnosis Steps
- 1. Identify the direction: Which side has no audio?
- 2. Test from different network: Does it work on cellular/different WiFi?
- 3. Check firewall logs: Are RTP packets being blocked?
- 4. Packet capture: Use Wireshark to see if RTP packets are sent/received
NAT/Firewall Fixes
Required Port Forwarding
Enable STUN
STUN helps devices behind NAT discover their public IP address.
Disable SIP ALG
Important: SIP ALG causes more problems than it solves. Disable it on your router.
How to Disable by Router Brand
Asterisk/FreePBX Fixes
PJSIP NAT Settings
; /etc/asterisk/pjsip.conf [transport-udp] type=transport protocol=udp bind=0.0.0.0 external_media_address=YOUR.PUBLIC.IP.HERE external_signaling_address=YOUR.PUBLIC.IP.HERE local_net=192.168.0.0/16 local_net=10.0.0.0/8 local_net=172.16.0.0/12
FreePBX Settings
- 1. Settings > Asterisk SIP Settings
- 2. Set External Address to your public IP
- 3. Add local networks to Local Networks field
- 4. Submit and Apply Config
Softphone Fixes
Zoiper/Linphone Settings
- ✓ Enable STUN:
stun.ipcomms.net - ✓ Enable ICE (Interactive Connectivity Establishment)
- ✓ Try UDP first, then TCP if issues persist
- ✓ Check microphone permissions
Still Having Issues?
IPComms support can help diagnose one-way audio issues. Contact us with your SIP traces.