VOIP service uses the internet, it is essential that networks are properly configured to handle our traffic. This document will cover the ideal network setup for VoIP and troubleshooting.
Network
A network is a group of computers and other devices, such as IP phones, linked together. In this section, we’ll cover the types of devices that make up a network.
Internet Connection
It is necessary to have a fast enough and reliable internet connection in order to get the best quality of service from VoIP. Before implementing VoIP, we recommend making sure your internet can handle the increased use of bandwidth.
Bandwidth
Bandwidth is the amount of data that can be transmitted in a fixed amount of time. This is expressed as an upload and download speed in Mbps (megabits per second), and lets you know how much data you have to work with. Since bandwidth is finite, the number of devices at the same location and on the same network, as well as concurrent phone calls, will impact the availability of bandwidth. Streaming and downloading both use large amounts of bandwidth. A lack of bandwidth can affect the sound quality of VoIP phone service, usually in the form of latency and choppy audio.
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You can check your speed here: http://www.speedtest.net/
The table below shows the minimum bandwidth required to make calls, as well as the recommended speeds for optimal performance.
| Number of Concurrent Calls | Minimum Required Bandwidth | Recommended speed |
|---|---|---|
| 1 | 100 Kbps Up and Down | 3 MBps Up and Down |
| 3 | 300 Kbps Up and Down | 3 MBps Up and Down |
| 5 | 500 Kbps Up and Down | 5 MBps Up and Down |
| 10 | 1 MBps Up and Down | 5-10 MBps Up and Down |
How Does VOIP Use My Bandwidth?
Modem (Standalone)
Typically provided by your ISP and connects you to the internet. Modems with built-in routers should be avoided. (See below)
Router
Very important for VoIP to work properly. It routes all the traffic within a local network (LAN) to external networks. It also provides security, allows you to share your network, and creates a private network for all connected devices.
Modem/Router
It’s not uncommon for ISPs to provide a modem with a built-in router, also known as a modem/router or “gateway”. These particular devices should be avoided as they are not always compatible with VoIP traffic and/or may require changes.
If there is a gateway, it is recommended to purchase a compatible standalone router. Then the gateway should be bridged to this router. This is done by disabling the routing portion of the gateway. The ISP should do this.
How does a phone call over VoIP work?
Now that we understand what a network is and the devices that make one up, let’s take a look at how a phone call works.
VoIP applications use two protocols, SIP (Session Initiation Protocol) and RTP (Real-time Transport Protocol). SIP is used for establishing and terminating a session (phone call). After the call is established, RTP takes over and is used for sending the voice data (packets) between phones. A packet is just a unit of data that carries information across networks.
/// Think of SIP as the stage manager. SIP prepares the stage for RTP by setting up its connection. Once RTP is finished with its stage (call), then SIP comes back to the stage to clean up after it. ///
Since these are different from protocols used in email (SMTP), streaming music and videos (RTSP), and browsing the internet (HTTP), oftentimes routers and firewalls are not configured by default to work with VoIP and the traffic is blocked or the voice packets will not be routed properly to our server and back.
SIP uses ports 5060 and 6060, and RTP uses ports 10000-49999. If these are blocked, the packets won’t be able to leave the network and ultimately get to where they need to be (our server). Think of a roadblock and the cars (packets) not being able to pass.
Troubleshooting
Broken Conversation
When the voice of either or both people on the phone cuts out during the conversation, sounds distorted or muffled, or there is a significant delay between when one person speaks and the other person hears the message. Most often associated with a problem with your internet connection or network setup...
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Lack of sufficient upload bandwidth
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High latency
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Competition from computers on the network uploading large amounts of data
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Incorrect settings on your network devices
Telephone equipment
Sometimes the phones themselves can cause sound issues. If one phone is having problems, perhaps try swapping out the piece of phone equipment (handset/headset). Interference from large and small electronic appliances can also cause sound quality problems. Crossed or tangled cable can cause feedback. For example, if your headset cable is crossed or tangled with your iPhone charger.
Static
Constant sounds like buzzing, low hum, crackles, pops, even when no one is speaking.
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Common causes of static are electrical interference from other devices. Move VoIP devices at least four feet away from other electronic devices.
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Static on only one phone could be the result of bad headset or handset.
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Check cables and connections. A defective telephone or ethernet cable or a bad port on your router can cause static.
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Regularly rebooting the modem and routers in the network is recommended to prevent issues such as static.
Issues typically related to network equipment/setup
Network equipment’s firewall settings can be a cause of sound issues if not set to correctly work with VoIP services. Not all modems, routers, and firewalls are VoIP compatible. Using incompatible devices, settings, or configurations can result in call quality or reliability issues. Refer below for recommended settings and compatible devices.
Double NAT (Double Routing)
A “double NAT” occurs when there are multiple routers on the same network doing network address translation. This is known to cause problems with VoIP applications. Ideally only one device is needed to perform routing functions. This scenario is most common when a user has a modem/router + standalone router. It is best to eliminate or bridge extra routers or modem/router combos on the network. It is recommended for the user to contact their ISP to bridge the modem/router device. If it is not possible to bridge the device, the user will need to exchange the modem/router for a standalone modem.
Phone not obtaining an IP address
If the phone is not obtaining an IP address, then it cannot connect to the network. Try rebooting the router first otherwise, this could be caused by:
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Ethernet cable not being plugged into the correct port on the phone (Ex. SW vs PC on Cisco)
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Phone not connected to the network properly
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DHCP not enabled
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Defective ethernet cable
If the customer is reporting the following symptoms, refer to the setting guidelines below for configuring their router’s firewall to properly handle our traffic.
One-Way or Two-Way Audio-
Dropped sound is when either or both people on the phone cannot hear the other person. Sound is lost during an active call and does not resume.
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Dropped audio on multiple phones is usually due to incompatible network equipment or the firewall settings of the router.
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Dropped audio on a single phone can be the result of faulty telephone equipment or the volume settings on the handset, headset, or speakerphone.
No Sound
There is no audio at all from the start of the conversation. Call rings and is answered but there is no audio.
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This can be the result of faulty telephone equipment or the volume settings on the handset, headset, speakerphone.
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It can also be caused by the phone’s connection to the network, either a bad connection or defective ethernet cable.
Router & Firewall Setting Recommendations
Most routers include a firewall, which is a filter that blocks traffic that should not be allowed in or out. For smaller offices with off-the-shelf routers, these firewalls may need to be modified or turned off. On enterprise-level equipment, certain rules may need to be added. The IT administrator should make these adjustments.
The following settings listed below can typically be found in the router’s web interface and can be easily checked on/off...
SIP ALG
SIP Application Layer Gateway is a setting that oftentimes prevents our traffic from flowing properly. If enabled, SIP ALG can cause various issues such as loss of connection with our servers, calls disconnecting, loss of one-way or two-way audio, phones continuing to ring after answered, and phones ringing randomly or out of sequence. Many routers will have the option to disable this feature, usually under the Firewall section (check the specific router’s user manual). If the option to disable SIP ALG is not available, we recommend purchasing a router that doesn’t have this feature or at least allows for it to be turned off.
It is recommended to have the customer’s IT administrator apply the following changes listed below to ensure it is done properly...
Ports
To ensure that traffic is not being blocked, the following ports used for VoIP communication to our servers should be opened:
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SIP Ports 5060, 6060, 5061, 6061
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RTP Ports 10000 - 49999
Other Router/Firewall Settings
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Make sure our IP is set as a trusted source: 72.1.46.0/23
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Increase UDP timeout settings up to 120
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Disable SIP Transformations
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Enable consistent NAT
SPI (Stateful Packet Inspection) / DoS Protection
SPI/DoS allows the router to approve or deny any information packets for security reasons. Oftentimes it will incorrectly identify our VoIP traffic as a security risk. DoS protection keeps track of how many connections are made to an individual web address and begins blocking access once a limit is reached. Our phones connect to the same site. The more phones in your network, the more likely SPI/DoS will begin to block connections. To prevent this, some routers will allow you to either disable SPI/DoS protection to allow more connections.
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