VoIP - Troubleshooting Guide

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VoIP Troubleshooting Guide

Our guide will help you fix common problems experienced by first-time VoIP users

Frequently Asked Questions & Common Issues




If you experience one-way audio or your number does not ring, even when your device says "Registered", you probably need to enable Port Forwarding on your internet router.

The Issue:

When a call comes through to our network for your number, we send that call through to you via the internet. If your device is connected directly to the internet, for example a mobile phone on 3G (using a softphone such as Zoiper), the call will come directly to your device. The softphone will ring and you will be able to hear the person on the other side.

If however your device is connected to the internet through a modem/router (eg. via Wi-Fi or network cable), the call from our network is sent to your modem/router (which is providing the internet connection) and not to your device directly, as your device is actually on a private network sitting behind your modem/router.

In order for your VoIP device to receive calls and all the associated media (audio) for those call, your modem/router needs to know that it must send this traffic to the VoIP device on your network.

If your modem/router has not been setup to do this, it will try to "guess" where it should send the VoIP traffic to. This might work sometimes but is very erratic and you may find issues with registering your device on our network and/or only receiving one-way audio on calls (you cannot hear the person on the other end).


The Solution:

You need to setup a "rule" on your modem/router so that it knows it should send all VoIP traffic to your VoIP device.

This "rule" is known as Port Forwarding.

Almost every DSL/Fibre/LTE router has a Port Forwarding section built into it. You would need to check your router's manual to determine how to access this interface and how to enable port forwarding for your particular router.

See examples of how to setup Port Forwarding in the How do I setup Port Forwarding section on this page.

You should also ensure that your VoIP device (whether that is a VoIP deskphone, computer with softphone or mobile device with softphone) has a static IP address on your network. For example: 10.0.0.5.

See examples of how to set your device to use a static IP address in the How do I set my device to use a static IP address on my network section on this page.

Once you have setup your VoIP device to use a static local IP address on your network and you have accessed your router's port forwarding interface, you then setup two port forwarding rules.

All traffic on ports 5060 (VoIP registration) and 10,000 to 20,000 (VoIP media/audio) should be forwarded to the LAN IP address of your VoIP device (eg. 10.0.0.5).


The instructions would differ depending on your device. Below are a few examples for common devices.

Examples

Example of how to set your Android device to use a static IP on your Wi-Fi:

Step 1. Go to settings and select Wi-Fi
Step 2. Tap and hold down on your current Wi-Fi connection
Step 3. Choose "Manage network settings"
Step 4. Tick "Show advanced options"
Step 5. Change "IP settings" from DHCP to Static
Step 6. Enter an IP address that is allowed on your network (eg. 10.0.0.X or 192.168.0.X)
Step 7. Enter your router's IP address under Gateway and DNS
Step 8. Tap "Save".





Example of how to set your iPhone device to use a static IP on your Wi-Fi:

Step 1. Go to settings and select Wi-Fi
Step 2. Tap on your current Wi-Fi connection
Step 3. Change the IP ADDRESS from DHCP to Static
Step 4. Enter an IP address that is allowed on your network (eg. 10.0.0.X or 192.168.0.X)
Step 5. Enter your router's IP address under Router and DNS





Example of how to set your Yealink Deskphone to use a static IP on your network:

Step 1. Login to your Yealink's web interface (check manual for details)
Step 2. Select the "Network" tab
Step 3. Change DHCP to Static IP Address
Step 4. Enter an IP address that is allowed on your network (eg. 10.0.0.X or 192.168.0.X)
Step 5. Enter your router's IP address under Default Gateway and DNS (optionally add 8.8.8.8, which is Google's Open DNS, under Secondary DNS)





Setting up Port Forwarding will be different for each modem/router, so you'll need to follow your manual on how to achieve this.


Example of how to setup Port Forwarding on a Netgear DSL router:

This example assumes that your VoIP device is listening on port 5060 for registrations, uses RTP (media) ports 10,000-20,000 and is using a static LAN IP of 192.168.0.11.

For a list of common VoIP phones and their default ports, please see the section "Port information for common VoIP phones" on this page.

Step 1. Login to your Netgear web interface (check manual for details)
Step 2. Select the "Advanced" tab at the top
Step 3. Under "Advanced Setup" choose "Port Forwarding / Port Triggering"
Step 4. Click "Add Custom Service"



Step 5. Enter a name of the SIP Registeration rule, select TCP/UDP and enter 5060 for Start and Stop Port.
Step 6. Then click Apply



Step 7. Repeat Steps 5 and 6 for SIP Media Ports 10,000 to 20,000



Step 8. Go back to Port Forwarding / Port Triggering and click "Add" at the bottom of the page



Step 9. Select the service name you created from the drop-down "Service" box and enter the IP address of your VoIP device and "Send to LAN Server"
Step 10. Click Apply.
Step 11. Repeat Steps 9 and 10 for the second service that was created (for ports 10,000 to 20,000).



You should now have port forwarding setup on the router to send all traffic on port 5060 and the port range 10,000 to 20,000 to your VoIP device directly.


Only one device should be registered on the network at a time.

The Issue:

While you can technically have several devices registered on the network for your number at the same time, the behaviour is perhaps not what you might expect.

If you register several devices on the network for your number at the same time, all the devices will register and be able to make outgoing calls. However while you may expect that all devices would then ring if an incoming call is made to your number, this is not the case.

The reason for this is due to the way VoIP works over the internet. When we receive an incoming call for your number our network then forwards the call through to the internet address of your device. If you connect from several different devices on different internet addresses (different internet connections) our network has no real way of knowing which internet address (IP address) to send the call to. Our network will then send the call to the last device that registered on the network.

VoIP devices on the network also have an automatic "re-registration" process that happens every few minutes and this time period is essencially random. Meaning that as each device re-registers automatically at random times, the "last registered" device will constantly change. So calls would end up randomly being sent to different devices.

Lastly if you are registering several devices on our network for your number and all these devices are connected behind the same internet connection (modem/router), the problem is further complicated by the fact that our network will send the traffic to your internet address, which is then received by your modem/router but your modem/router would have no way of being able to send that same traffic to all the devices. It will then try to "guess" which device it should send the call to. Leading to random behaviour in terms of receiving calls and in some cases also to "one-way audio" (see Question 1).


The Solution:

Should you wish to use several devices on the same number you need a PBX/Switchboard.

The PBX then registers on our network for your number and all your devices register directly to your PBX, as "extensions".

In this way, when our network receives a call, it sends that traffic to your PBX (which is the only device registered on our network for your number) and your PBX then has the built in function to send the call to all your extensions. Of course with a PBX you can change this behaviour to suit your needs, such as setting up an IVR (Interactive Voice Reponse) system, call queues, call forwarding, ring groups (ring certain extensions) and much more.

You can either buy a physical PBX/Switchboard installed on-site or use a Hosted PBX service (recommended). For more information on our Hosted PBX offering, click here.

The PBX registers on our network and all calls are sent/received between our VoIP network and the PBX only. You can then setup several "extensions" on the PBX itself and your multiple VoIP devices then connect to the PBX (ie. not our VoIP network). When setting up the extensions on the PBX you tell the PBX to communicate with each extension using a different "port".

For example, if you had a mobile phone with Zoiper, a PC with a VoIP program and a normal VoIP deskphone in your network:

- You can setup an extension, say 9000 on the PBX and tell it that this extension is "listening" on port 9000.
- You can setup another extension, say 9001 on the PBX and tell it that this extension is "listening" on port 9001.
- You can setup another extension, say 9002 on the PBX and tell it that this extension is "listening" on port 9002.

You could then use extension 9000 on the mobile phone (telling the softphone it should "listen" on port 9000), use extension 9001 on the PC (also telling the PC's software to "listen" on port 9001) and finally use extension 9002 on the VoIP Deskphone and telling the deskphone it should "listen" on port 9002.

The final step would then be to setup Port Forwarding on your internet router (see Question 1 above) where you forward traffic received on port 9000 the the IP of your mobile phone, 9001 to the IP of your PC and 9002 to the IP of your VoIP deskphone.

These steps can simply be repeated for any number of extensions you may add.


This can be caused by poor internet quality, bad codecs or a faulty device.

VoIP (Voice over Internet Protocol) itself is a digital service, it does not drop or break-up calls or suffer from any added noise/echo problems. However as the service is supplied over the internet, it is dependant on a stable connection with quality bandwidth. The device you use with your VoIP service may also determine the quality of the voice audio.

If your internet connection drops (even for a second) it can drop a call that is ongoing at the time. If your ISP is having issues with their bandwidth or you are being shaped by the ISP, this too may effect the quality of the voice audio being sent/received during a call. If your internet connection is too slow you will experience break-ups in the audio or a delay in the sending and receiving of audio (latency).

It is for this reason that we recommend a minimum of 2Mbps internet speeds on unshaped data from your ISP.

It is also a good idea to check if your router supports QoS (Quality of Service), which is a feature you can enable to ensure that your router will always give first priority on your network to VoIP traffic over any other usage.

If you are certain the your internet connection is of suffcient speed and quality and you still experience break-up or poor quality audio, you may want to check which codecs are enabled on your device.

A codec, which stands for coder-decoder, converts an audio signal (your voice) into compressed digital form for transmission (over VoIP) and then back into an uncompressed audio signal for replay. Sometimes using a different codec can improve voice quality as different devices work better using diffent codecs. We recommend having a.Law, u.Law and GSM enabled as standard. You may also look at purchasing a license for the G.729 codec which offers the best compression vs performance for most devices.

If after trying different codecs there is still no improvement, the problem might be with the device itself, try another device to test.


Below is a PDF document with step by step instructions.




Below are intructions on how to setup a Yealink Deskphone.

Example of how to set your Yealink Deskphone to use a static IP on your network:

Step 1. Login to your Yealink's web interface (check manual for details)
Step 2. Select the "Network" tab
Step 3. Change DHCP to Static IP Address
Step 4. Enter an IP address that is allowed on your network (eg. 10.0.0.X or 192.168.0.X)
Step 5. Enter your router's IP address under Default Gateway and DNS (optionally add 8.8.8.8, which is Google's Open DNS, under Secondary DNS)




Example of how to add your VoIP account to your Yealink Deskphone:

Step 1. Login to your Yealink's web interface (check manual for details)
Step 2. Select the "Account" tab
Step 3. Select "Account 1" from the "Account" drop-down box and select "On" for "Account Active"
Step 4. Enter your VoIP number as supplied in the activation email for "Label", "Display Name", "Register Name" and "User Name"
Step 5. Enter your VoIP password as supplied in the activation email for "Password"
Step 6. Enter "sip.nexus.co.za" (port 5060) in the "SIP Server" field
Step 7. Optionally you can "Enable" the Outbound Proxy Server and enter "sip.nexus.co.za" (port 5060) in the "Outbound Proxy Server" field.
Step 8. Click the "Confirm" button at the bottom of the page. The "Accounts Status" should change to "Registered" once successfully connected



You should now be able to make and receive calls on your new Yealink deskphone.

Below is a list of common VoIP phones and the default ports used by the device. This can assist you when setting up Port Forwarding.


Xten softphones

Port Type Number Service
UDP 5060/5061 SIP COMMUNICATIONS (plus custom ports)
UDP 5082 SIP COMMUNICATIONS (OUTBOUND PROXY)
UDP 8000 . 8012 RTP, RTCP, VOICE

Two additional ports after 8001 are required for each additional line used. For example, if using a second line, UDP ports 8002-3 will be used.

Linksys Range of phones/Adaptors

Port Type Number Service
UDP 5060/61 SIP COMMUNICATIONS (plus custom ports)
UDP 5082 SIP COMMUNICATIONS (OUTBOUND PROXY)
UDP 49152-65534 RTP,RTCP,VOICE

Sipura Range of phones

Port Type Number Service
UDP 5060/61 SIP COMMUNICATIONS (plus custom ports)
UDP 5082 SIP COMMUNICATIONS (OUTBOUND PROXY)
UDP 16384-16482 RTP,RTCP,VOICE

SNOM Range of phones

Port Type Number Service
UDP 5060/61 SIP COMMUNICATIONS (plus custom ports)
UDP 5082 SIP COMMUNICATIONS (OUTBOUND PROXY)
UDP 49152-65534 RTP,RTCP,VOICE

Flexor 151 Adaptor

Port Type Number Service
UDP 5060/5066 SIP COMMUNICATIONS (plus custom ports)
UDP 5082 SIP COMMUNICATIONS (OUTBOUND PROXY)
UDP 5004 RTP,RTCP,VOICE

Grandstream Range of Products

Port Type Number Service
UDP 5060/61 SIP COMMUNICATIONS (plus custom ports)
UDP 5082 SIP COMMUNICATIONS (OUTBOUND PROXY)
UDP/TCP 5004 RTP,RTCP,VOICE

Cisco Products

Port Type Number Service
UDP 5060/61 SIP COMMUNICATIONS (plus custom ports)
UDP 5082 SIP COMMUNICATIONS (OUTBOUND PROXY)
UDP/TCP 16384 to 32768 RTP,RTCP,VOICE

Asterisk servers

Port Type Number Service
UDP 5060 SIP COMMUNICATIONS
UDP 4569 IAX2 PROTOCOL
UDP 5036 IAX PROTOCOL
UDP 10000-20000 RTP MEDIA STREAM
UDP 2727 MEDIA GATEWAY CONTROL

Siemens Range of phones/Adaptors

Port Type Number Service
UDP 5060/61 SIP COMMUNICATIONS (plus custom ports)
UDP 5082 SIP COMMUNICATIONS (OUTBOUND PROXY)
UDP 5004-5020 RTP,RTCP,VOICE

Yealink Range of Products

Port Type Number Service
UDP 5060/65 SIP COMMUNICATIONS (plus custom ports)
UDP 5082 SIP COMMUNICATIONS (OUTBOUND PROXY)
UDP 11780-11800 RTP,RTCP,VOICE

Hosted Unified Comms . Telepo Softphone

Port Type Number Service
TCP 5060/61 SIP
UDP 49152-65535 RTP,RTCP,VOICE