This article explains what to do if you experience one of the following issues:
a. You can’t make or receive any calls: The phone is not connecting to the service at all.
b. One-way audio: The call connects, but only one person can hear the other.
c. No audio on both sides: The call connects, but neither person can hear anything.
Before you start: Do a quick test
Before changing any settings, check whether the issue is related to your network.
Connect your desk phone to a different network, such as a mobile hotspot.
Try making and receiving calls.
If calls work, the issue is likely related to your network or firewall settings.
If calls don’t work, the issue may be related to your phone configuration or account.
If you can’t make or receive any calls
This issue is often caused by a local network conflict. You can try changing the local SIP port.
Step 1: Log in to your desk phone
On the phone, go to Menu > Status to find its IP address.
Open the IP address in a web browser (for example, past the IP address <phone_IP_address> after the https:// into the address bar).
3. Log in using admin credentials:
Check the label on the phone
Or try admin / admin
Step 2: Open SIP port settings
Go to Account → Register → Advanced
Locate the Local SIP Port setting
Step 3: Change SIP port settings
Change the port from the default (usually 5060) to another value, such as:
5062
5068
5080
2. Save the changes
3. Test by making a call
If you have one-way audio or no audio
This issue is usually caused by firewall restrictions or RTP port configuration.
Step 1: Log in to your desk phone
On the phone, go to Menu > Status to find its IP address.
Open the IP address in a web browser (for example, past the IP address <phone_IP_address> after the https:// into the address bar).
3. Log in using admin credentials:
Check the label on the phone
Or try admin / admin
Step 2: Open RTP settings
Navigate to voice or network settings, such as Settings → Voice → RTP
(menu names may vary by model)
Step 3. Change the RTP port range
Update the port range:
a. Min RTP Port: 20000
b. Max RTP Port: 22000
2. Save the changes
3. Test with a call
Advanced troubleshooting (network / firewall)
If the steps above do not resolve the issue, the problem is likely related to your network configuration.
Contact your network administrator and ask them to review and adjust firewall settings to ensure proper operation of desk phones.
Required ports for desk phones
Below is an overview of the ports commonly used by desk phones:
Protocol | Port Number | Transport | Description |
SIP | 5060 | UDP / TCP | Used for call signaling (starting, ending, and managing calls). |
Secure SIP (TLS) | 5061 | TCP | Encrypted SIP signaling (used in some configurations). |
RTP / RTCP | 10000–20000 (or configured range, e.g., 20000–22000) | UDP | Used for transmitting audio during calls. |
HTTPS | 443 | TCP | Secure provisioning and web interface access. |
HTTP | 80 | TCP | Web interface access (non-secure, optional). |
NTP | 123 | UDP | Time synchronization (important for logs and certificates). |
TFTP | 69 | UDP | Used for auto-provisioning (if applicable). |
Critical: Disable SIP ALG
This is the most important step.
Many routers include a feature called SIP ALG (or SIP Helper) that modifies SIP traffic and often breaks VoIP functionality.
Action:
Locate SIP ALG in your router/firewall settings (commonly under Security, Advanced, or NAT)
Disable it completely
Save changes and reboot the router if needed
Allow outbound traffic
Ensure the firewall allows outbound traffic for:
SIP: 5060 (UDP/TCP), 5061 (TCP)
RTP: Configured UDP range (e.g., 10000–20000 or 20000–22000)
If you changed the RTP range on the phone, make sure the firewall allows the same range.
Optional: Configure port forwarding
Port forwarding is not required in most cases and should only be used in restrictive network environments.
If needed:
Forward SIP port (e.g., 5060) to the phone’s local IP
Forward RTP port range (e.g., 20000–22000) to the same IP
Security Note: For enhanced security, if the VoIP provider has a static IP address or range, you should set that as the Source IP for these rules. This ensures that only traffic from your provider is allowed.
Adjust UDP timeout period
Firewalls often close UDP connections after a short period of inactivity (e.g., 30 seconds). This can cause the phone to lose its registration with the provider.
Action: Find the UDP timeout setting in the firewall and increase it for the SIP port. A value of 180 seconds or higher is recommended to ensure the connection remains active.
Enable STUN on the phone
This step helps the phone understand its own network environment and communicate correctly through your firewall.
What to do: Log into the web interface of your desk phone.
Action:
Navigate to the Account > Register page.
Find the setting called NAT Traversal.
Change its value from “Disabled” to “STUN”.
In the STUN Server field that appears, enter: stun.yealink.com
Click Confirm to save the changes. The phone will apply the settings and may reboot.
Why: STUN helps the phone discover its public IP address and port, which is crucial for establishing the audio path correctly when it’s behind a firewall.