One-Way Audio on Calls — How to Diagnose and Fix

Introduction

One-way audio means one party on a call can hear the other, but not the other way around. For example, the caller can hear you but you cannot hear them, or vice versa. This is almost always a network issue rather than a PBX configuration problem.

Understanding why it happens

SIP calls use two separate communication channels:
- SIP signalling — handles call setup and teardown (usually port 5060)
- RTP media — carries the actual audio (uses a range of UDP ports, typically 10000–20000 on Yeastar)

One-way audio usually means the SIP signalling is working (the call connects), but RTP audio is being blocked in one direction by a firewall or NAT issue.

Most common causes

1. SIP ALG enabled on the router

SIP ALG (Application Layer Gateway) is a feature on many consumer and SMB routers that tries to "help" with SIP traffic — but often breaks it instead.

Fix: Log into your router admin page and disable SIP ALG. Look for it under Firewall, NAT, or Advanced settings. The exact location varies by router brand.

2. RTP ports blocked by a firewall

Your firewall may allow SIP signalling (port 5060) but block RTP audio ports.

Fix: Ensure UDP ports 10000–20000 are open outbound from the phones/PBX to the internet. If you have an on-premises firewall (not just a router), check the outbound rules.

3. NAT traversal issues (on-premises phones calling through Cloud Edition)

Cloud Edition handles NAT automatically for registered phones. However, if the phone's SIP settings have incorrect NAT options configured, the phone may send RTP audio to its private IP address instead of its public IP.

Fix:
1. In the phone's web interface, go to Account > Advanced (or NAT).
2. Ensure NAT Traversal is enabled.
3. Ensure STUN is not set to a defunct server.

4. Codec mismatch

If the PBX and the remote party (your SIP carrier) don't share a compatible audio codec, the call connects but audio may not flow.

Fix:
1. Go to Extension and Trunk > Trunk, edit your SIP trunk.
2. Click the Codec tab.
3. Ensure G.711 a-law (PCMA) is listed first — this is the standard codec for UK SIP carriers.
4. Move G.711 a-law to the top of the list using the arrow buttons.
5. Click Save and Apply and test the call again.

Quick test: is it the phone or the network?

Make a call using the Linkus Web Client from a browser (see the Linkus Web guide). If audio works in Linkus but not on the desk phone, the issue is with that specific phone's network configuration.

Tips

• One-way audio that only affects calls to/from specific carriers (but not all calls) usually points to a codec issue with that carrier.
• One-way audio on ALL calls from a site usually points to SIP ALG or a firewall blocking RTP.
• After making network changes, make a test call immediately — don't wait until a customer complains.

Summary

Disable SIP ALG on your router first — this fixes the majority of one-way audio issues. If that doesn't help, check RTP port access and codec configuration.