Grandstream UCM & IP Phone One-Way Audio / NAT Traversal Fix
Fix one-way audio where caller hears you but you cannot hear them (or vice versa) on Grandstream UCM IP telephony setups.
⚡ Direct Answer & Quick Diagnostic Summary (AEO)
To resolve one-way audio where one party cannot hear voice streams on Grandstream UCM6200/6300 IP-PBX phone systems, disable SIP ALG (Application Layer Gateway) on the office edge router and firewall. SIP ALG corrupts Session Description Protocol (SDP) media payload headers. In Grandstream UCM web administration, navigate to PBX Settings → SIP Settings → NAT. Set NAT Mode to "External IP" (if using a static IP) or "Keep-Alive" with STUN (stun.l.google.com:19302), enter your Public WAN IP, and define all internal subnets (e.g., 192.168.1.0/24). On the gateway router, create a virtual server port forwarding rule for RTP Voice Stream ports: Protocol UDP, Port Range 10000–20000, pointing directly to the UCM's static IP. In PBX Codec settings, prioritize PCMA (G.711 A-law) as Codec 1 and PCMU (G.711 U-law) as Codec 2 to eliminate transcoding negotiation failures across Pakistani telecom SIP trunks.
Primary Root Causes
Can't resolve this issue or hardware failure halted your shift?
Our certified field hardware & network engineers are on standby for emergency on-site repair across Korangi, SITE, Clifton, DHA, Shahrah-e-Faisal, and all Karachi commercial areas.
Step-by-Step Diagnostic Checklist
Step 1: Disable SIP ALG on ISP Router & Firewall
Log into your main fiber router and firewall. Navigate to Security / ALG Settings and set SIP ALG (Application Layer Gateway) to Disabled. SIP ALG routinely alters SIP header ports, breaking audio return paths.
Step 2: Configure Grandstream UCM NAT Settings
On UCM web interface, go to PBX Settings → SIP Settings → NAT. Set NAT Mode to "External IP" (if you have static IP) or "Keep-Alive" with STUN. Enter your External IP Address and specify Local Network Subnets (e.g. 192.168.1.0/24).
Step 3: Forward RTP Voice Stream Ports
On your router firewall, create a Port Forwarding rule for RTP Media Ports: Protocol UDP, Port Range 10000-20000, Forward to Grandstream UCM IP Address.
Step 4: Configure STUN Resolution on Remote IP Phones & Wave App
In Grandstream Wave Mobile App or remote IP phone settings, set NAT Traversal to STUN and enter stun.l.google.com:19302 as the STUN server.
Step 5: Standardize Audio Codecs across SIP Trunks
Go to PBX → SIP Settings → Codecs. Ensure PCMA (G.711 A-law) is set as Priority 1 and PCMU (G.711 U-law) as Priority 2 across all trunks and extensions to eliminate codec transcoding drops.
Frequently Asked Questions
Why can I hear the caller but they cannot hear me on Grandstream PABX?
One-way audio is almost always caused by SIP ALG enabled on your internet router or RTP audio ports (UDP 10000-20000) not forwarded to the Grandstream UCM static IP.
What is SIP ALG and why should it be disabled for VoIP?
SIP ALG is a router feature intended to help VoIP, but in practice it rewrites SIP packets incorrectly, causing dropped calls, failed registrations, and silent/one-way audio.
Need Enterprise Deployment or Support for PABX & IP Telephony?
If your team needs full hands-on setup, SLA monitoring, or emergency on-site assistance in Karachi, our certified engineers are ready.