What is VoIP?
Voice over IP (VoIP) is a methodology and group of technologies for the delivery of voice communications over IP networks (such as the Internet). VoIP can be thought just like traditional telephony - copper wires allowing for voice communication over the Public Switched Telephone Network (PSTN), but instead of using copper wires, VoIP uses the Internet.
Callture VoIP Architecture
Figure 1.1 shows a typical network architecture for the Callture Cloud-based phone system. It consists of the following components:
PSTN - This is the Public Switched Telephone Network, or where trunks and phone numbers connect to each other. Anytime a phone call is made to the outside world, or to a phone number, it travels across the PSTN.
Callture PBX - This is the Callture Cloud-hosted Phone System that handles all of the phone switching, inbound/outbound calling, voicemail, and IVR functionality.
Modem - The modem is typically provided by an Internet Service Provider (ISP) and allows them to terminate an Internet connection (such as cable, DSL, or fiber) to a residential or business location. The modem may also be a combination device that acts as a firewall or provides WiFi capability in addition to terminating the Internet connection.
If a separate Router/Firewall is being used in conjunction with the modem provided by the ISP, it is always best to put the ISP’s modem in ‘bridge mode’ so that all Internet traffic flows directly to the Router/Firewall. The ISP that supplies the modem should be contacted for instructions on how to set their modem into bridge mode.
Router/Firewall - Most businesses have a router/firewall in addition to the ISP provided modem. This device serves many functions. The router functionality separates external networks (also known as the Wide Area Network (WAN), or the Internet) with internal networks (also known as the Local Area Network (LAN)). The firewall functionality filters Internet traffic and blocks unknown or malicious connections, but allows traffic that is explicitly configured to pass through the firewall. Often, the router/firewall will also provide WiFi capability.
In some cases, the modem provided by the ISP also serves as a router/firewall negating the need for a separate device.
Switch - a network switch is used for connecting computers and devices together on the Local Area Network (LAN). Depending on the business size, many network switches may be connected together to handle the number of devices on the network. When Voice over IP is being used on the LAN, it is recommended that the switch have Power over Ethernet (PoE) functionality. PoE functionality provides power to VoIP phones through the network wiring providing two main benefits - first, the phones do not need a separate power adapter at each desk, and second, the entire PoE network switch can be placed on a battery backup device (UPS) keeping them on in the event of a power outage.
Phones - VoIP phones come in two main flavors. There are desktop phones (also known as hard phones) and softphones. Desktop phones are traditional telephony devices and typically have a display, a number pad, and a handset. Most desktop phones also have a secondary Ethernet port (also known as a passthrough port) that you can also connect your computer to (see Figure 1.2 below). This eliminates the need to run separate Ethernet cabling to each phone.
The other type of VoIP phone is a softphone. A softphone functions much like a desktop phone, but it is a software program running on a computer. The softphone uses a computer’s microphone and speaker devices to send and receive voice communications.
VoIP Network Requirements
How can you determine if the network is ready for VoIP?
The first thing to do is to estimate the maximum number of concurrent calls. This is not the total number of phones connected to the Callture Cloud-based phone system, but rather the anticipated number of phone calls in progress during peak periods. In many cases, this can be estimated as a simple ratio of phones to phone calls.
Normal businesses will have a call ratio of phones to phone calls somewhere between 3:1 and 4:1. A 4:1 ratio means that for every 4 phones in the company, one of those phones will be on an active phone call. As an example, a company with 12 phones and a 3:1 call ratio will have approximately 4 people on the phone during peak periods. If that same 12 phone company estimates a 4:1 call ratio, approximately 3 people will be on the phone during peak periods.
Businesses with lower call ratios such as doctors offices and manufacturing facilities may have a call ratio of 5:1 or 6:1 (or less).
Businesses with higher call ratios such as call centers may be closer to 1.5:1 or 2:1 during peak periods.
Once the approximate number of concurrent calls has been determined, the amount of bandwidth required can be estimated.
Each concurrent phone call in or out of the Callture Cloud-based phone system utilizes 40 Kbps worth of upload and download bandwidth. Multiplying the number of concurrent calls by 40 Kbps will give an estimate of the total amount of bandwidth required. For instance, a business with an estimated 5 concurrent calls will need to have 200 Kbps worth of available bandwidth for their VoIP communication.
Internet Speed Testing
Running a speed test can help determine how much Internet bandwidth is available. A good speed test to use can be found at Speedtest.net. Open a browser, navigate to Speedtest.net, and click the ‘Begin Test’ button.
When the test finishes, the measured results are displayed (Figure 2.2).
Download speed is the available bandwidth coming into the network, and upload speed is the available bandwidth leaving the network. Download speed will typically be larger than upload speed, and the lower of the two numbers should be used to determine if the estimated number of concurrent calls that will be compatible with the amount of bandwidth available.
In the Figure 2.2 example, the upload speed is 4.25 Mbps (Megabits per second). The previous example of 5 concurrent calls required an estimated 200 Kbps (Kilobits per second). Multiplying 4.25 Mbps X 1000 will convert the available bandwidth to 4250 Kbps. The 5 concurrent calls required will use up approximately 5% of the total bandwidth available, which means that this Internet connection has plenty of bandwidth for VoIP communication.
One thing to keep in mind is that if the Internet connection is a shared voice and data connection, the total amount of bandwidth is good to know, but the amount of available bandwidth is also important. If the Internet connection is saturated with data traffic, there may not be enough bandwidth left over for VoIP communication. This issue can be mitigated by utilizing Quality of Service (see Quality of Service below).
Internet Quality Testing
In addition to Internet bandwidth testing, it is also important to measure the quality of the Internet connection. Even with a good amount of bandwidth available, other factors such as jitter and packet loss can have negative effects on voice quality.
The combined measurement of Internet speed and quality is known as MOS (Mean Opinion Score). MOS provides a numerical measure of the quality of human speech over a VoIP line. MOS is a 1 to 5 scale of VoIP quality, 1 being the worst, and 5 being the best. A score of 4 or higher is great for VoIP.
A good test for measuring MOS can be found at http://myspeed.visualware.com/index.php. Open that URL in a web browser, and set the default test type to ‘VoIP’ as shown in Figure 2.3. It is best to run this test multiple times during an average business day, and especially during peak traffic periods.
Next, select the appropriate ‘green’ country and state/province that is closest to where the test is being run (see Figure 2.4 below).
In the window that pops up, select the closest test location. (see Figure 2.5 below).
If a Java window pops up, click ‘Run’ (see Figure 2.6 below). Java must be enabled for this test to function properly.
When the test appears, select ‘G.729’ from the ‘Codec’ drop-down box and select the number of lines from 1 to 5 (see Figure 2.7 below). This should be equal to the number of concurrent calls anticipated. If the number of anticipated concurrent calls is greater than 5, select the maximum value of 5.
Once selected, click the ‘Click to start test’ button. After the test runs, the results will be displayed.
Green lights for Jitter and Packet Loss with a MOS of 4 or greater means that the network is well suited to VoIP communication.
If any warnings are displayed for either jitter or packet loss, or if the MOS is below 4, the network should be inspected by a qualified IT administrator to determine the source of the issue.
A high amount of jitter often means that the network is congested, and typically this is due to a bottleneck at the Internet connection. This condition can be mitigated by implementing Quality of Service (see Quality of Service below).
A high percentage of packet loss means that data packets are being lost from point A to point B. This is measured as an overall percentage of lost packets. 0% packet loss is ideal, however 1-2% packet loss is acceptable and often unnoticeable. Many customers with large amount of packet loss do not notice this issue until they implement VoIP because standard data traffic (such as web surfing, or watching YouTube videos) can re-send data packets and recover from the losses. Voice communication on the other hand is real time, and packet loss is perceived as clipping and audio drop-outs. Packet loss issues should be documented and reported to the ISP for investigation and resolution.
Quality of Service
Quality of Service (QoS) is the process of prioritizing one type of network traffic over another. VoIP network traffic should be prioritized over all other types of traffic since it consists of real time communication. QoS is most important at the Router/Firewall level since this is often a bottleneck in terms of the amount of available bandwidth. For instance, a typical LAN will have Gigabit (Gpbs - also measured as 1000 Mbps, or Megabits per second) switches, but an Internet connection is something like 60 Mbps download, 5 Mbps upload (or 1000 Mbps on the LAN vs 60 Mbps at the Internet connection).
Different brands of firewalls implement QoS differently, however it is typically either a per device, or per port setting. If QoS is implemented per device, each phone should be given the highest priority. If QoS is implemented per port, the standard VoIP ports should be given the highest priority. These ports are UDP 5060 and UDP range 10,000 through 20,000. A qualified IT administrator should be consulted to properly implement QoS in the firewall.