Troubleshooting Single audio in VoIP and solving it
(Single audio is also known as One-way audio) Single audio, where one side or one party can hear the other but not reverse, is typically indicative of something stopping either the outbound or inbound audio from reaching the receiving party. Single audio can be caused by network issues, NAT (Network Address Translation) issues, SIP or VoIP capabilities that are still switched on on the modem or firewalls, so finding the cause will require simplification of the connection by the elimination of some equipment, then testing. Once the problem device is found it's pretty much down to initiating changes that resolve the issue. Here is how to troubleshoot Single audio.
1) Check your equipment.
The first thing that you need to eliminate is a faulty phone, handset earpiece or a headset on a softphone.
To do so, do the following:
Softphone
- Check that you have audio in both the earpiece and microphone.
- Verify that your microphone is connected properly. Try using Windows Sound Recorder (or another application) as a check and make sure that the microphone is selected as the input device in your Control Panel.
- Make a test call to yourself and start speaking to verify the microphone meter is showing movement.
Regular phone
- Sometimes the best testing can be achieved using a simple corded phone. Connect one that is working correctly, both the mouthpiece and earpiece.
IP Phone
- Make sure the phone is functioning correctly.
2) Simplify the connection.
Now that you know the equipment (phone/softphone with headset) is functioning correctly it's time to simplify the LAN (Local Area Connection) and the way the VoIP device is connected. The first step in one way audio troubleshooting is to simplify the connections. This allows you to identify the actual cause of the VoIP one-way audio.
Do the following:
Connect the VoIP ATA, IP Phone, or PC with softphone directly to the modem device. Make sure you get registered and obtain a valid IP address. In many cases, this could be a public IP address. Make a test call. If single audio still exists check to see if you have a public or private (192.168.1.xxx) IP address.
Public IP
Call your VoIP provider. If you are getting one-way audio with a public IP address, there is an issue with the way the VoIP provider is handling the call.
Private IP
If you are getting a private IP address from your modem it is acting as a NAT router and in this case preventing the audio from making it to the ATA. Check for SPI (Stateful Packet Inspection) or any firewall settings that may be the cause. Turn them off and retest.
If you are getting a private IP address and not able to get two-way audio, you will want to set the modem to bridge mode and allow your router to act as the only NAT router on the network. Instructions to do this should be in the manual of your modem.
3) Reconfigure your LAN network.
After completing step two you should have achieved a good two-way audio call.
Now you will want to configure your network for your VoIP to work correctly. This typically might include eliminating a second layer of NAT from your network design.
DSL/Cable modem as router
More and more modems are acting as routers (both hard-wired and wireless), firewalls and handing out DHCP. If you have to keep your modem as the primary NAT router (perhaps you need the wireless capabilities) then you will need to eliminate any secondary router from creating a second layer of NAT.
Use a switch on the LAN in place of a router or disable DHCP on the second router.
Bridge the modem
Set the modem to bridge mode and allow the public IP address to sit on the WAN side of your router. This may require placing PPPoE settings in your router. Doing this will make sure that the router handles all NAT translations.
Two routers
If you have two routers, perhaps one wired and the other wireless you will need to configure these routers correctly for best network practice.
Port forwarding
Configuring port forwarding on your router can be beneficial and resolve many NAT related single audio issues. After completing the best practices with your network you might use port forwarding as a way to make sure that the router knows where to send incoming VoIP audio.
You can try to allow all RTP traffic through - UDP ports 10000 to 20000 on your firewall.
This does not pose any security issues.
4) Make sure the modem/router does do its NAT Function well and/or disable any SIP/VoIP functionality in the modem.
If you still can't get it to work properly, do start a support ticket so that we can help.
Do know that there are several ways to solve this and that, up till now, the issues have never Not been solved.
So don’t worry, we’ll get this sorted
Extra information that may help to solve local issues that may be causing you to experience single-audio.
Avoid one-way voice issues using Port Forwarding
One way voice issues can occur because the router of the network doesn't know where to send the voice packets or "forgets" where to send the packets.
Port forwarding gives a static route for the packets to follow and this way they should always arrive at the correct device.
To port forward on the local network you will need to take a few steps:
- Assign static IPs to all of the IP phones on which you wish to port forward
- Enable port forwarding on the local router and forward the set ports to the correct static IP
- Assign RTP port ranges to individual phones
- Port forward the RTP Port ranges to the correct endpoint
Assigning Static IPs
You will need to assign a static IP to both the router and each IP phone.
Most routers have a different interface and so you should use the following site to determine exactly how to port forward on your specific make and model: https://portforward.com/router.htm
The process itself should be similar on most routers.
Router Configuration
- Start up your browser, such as Chrome or Firefox
- Enter your default Gateway, otherwise known as the routers IP, this can be found by opening a command prompt and entering the command "ipconfig"(windows) or "netstat -nr" if you are on Linux or Mac.
- When you go to the default gateway, most likely you will be prompted for a username and password.
This can either be found on the bottom of the router or if it is an older model you may be able to find the default credentials, by clicking here and finding your router's model. - From now on the process may well be different depending on your router's make and model, however, you will need to find a tab or screen which shows you a list of current devices currently on the network.
You should be able to see the IP phone's names and any other devices. - It is advisable to have pen and paper ready for the next step so you can write down the IP, Phone and SIP Port and RTP Port
- In the list available devices on your network, go to the first entry which is an IP Phone (the hostname will usually be some indication of the device) and edit the entry.
In the device's option, there will be an option which states how the IP address was obtained and will usually be DHCP.
Now: note down the name of the phone, and something which allows you to remember it, and it's IP number.
Set the option of how to obtain IPs to "Static" and then input the existing IP into the IPv4 Configuration.
If you are required to enter a subnet mask then this will usually be 255.255.255.0
SIP Port Forwarding
Now you will need to find the "Port Forwarding" page, following the guide above, and add a new entry for the IP you have just set.
You will need to set a start port and an end port.
Typically, SIP uses the port 5060, and it will also use the port above that so 5061, both of these are used for Data and Control.
Therefore you will need to leave a free logical port in between, each IP Phone.
So for example, the first phone you port forward might be set to port 5062, however, in port forwarding, you will have to specify the start port of 5062 and then end port of 5063.
Use the website we mentioned above, to aid you in finding these settings on your router.
The protocol of this port forward should be UDP and TCP.
RTP Port Forwarding
You will also need to add a port forward for the RTP ports.
Again, this has to be a range of ports and have a start port and an end port.
However, for RTP there needs to be an available range which is relative to the number of phones you have on a network.
For example, if you have 10 phones in an office then you will have 20 free logical ports in the network, however, if you have 50 phones in an office then you will need to have 100 free logical port.
Generally, the more phones you have the larger the range, however, it is not a lateral increase.
An actual example would be, if you have 13 phones in an office then for one of the phones there may be a range of 16500 - 16520 as there is a range of 20 free ports.
The protocol used for the RTP ports should be UDP.
You should now have at least two entries for each IP, depending on the router.
However, you should have the following:
- 1 Entry for the SIP range e.g. 5062-5063 on TCP/UDP
- 1 Entry for the RTP range e.g. 36500-36520
- The default RTP Range is: 16384-65000 Try not to go out of this range unless needed.
Phone Configuration
After port forwarding, note down the port range you used, and in a new tab on your browser, go to the phone's IP which you should have noted down.
You will be prompted for a username and password, which if unchanged can be found by checking the relevant manual or by googling the phone's make and model.
- After logging in, go to the network page on your phone, and you should see options for the RTP Options and the NAT/Local IP Settings
- In the RTP settings, input the port range that you set earlier, i.e. 16500-16520
- In the NAT/Local IP Settings, input the IP you set the phone and the port range which you set, i.e. 5062 and 5063
- You will now need to repeat this for each phone in the network.
Remember though, the process differs slightly for each router/phone brand and some screen or names may be slightly different but don't let this discourage you.
Last updated: 01/01/2024
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