What is latency?

Latency refers to the time it takes a voice packet to reach its destination. Latency is measured in milliseconds (ms)—thousandths of a second. Latency of 150ms is barely noticeable and generally acceptable. Latency higher than 150ms adversely affects VoIP QoS, while latency higher than 300ms is generally unacceptable.

What is jitter?

Jitter measures the variation of packet arrival times—or simply put, how much latency varies within the network. Jitter is often caused by network congestion, timing drift, or route changes. Jitter is measured in milliseconds (ms)—thousandths of a second. Jitter greater than 50ms can increase latency and result in packet loss.

What is Packet Loss (also known as data loss)?

Packets are sent over the Internet and reassembled at their destination. Packet loss occurs when some packets are dropped by congested network routers or switches, or discarded by the jitter buffer. If you miss one out of every 10 words, or 10 words all at once, chances are you won’t understand the conversation.

1. Upgrade your Internet connection.

VoIP QoS depends on the upload and download speed of your Internet connection. Speak with your Internet service provider (ISP) if you've experienced any VoIP call quality issues. Many ISPs will upgrade your connection speeds for free simply to keep you as a customer. Otherwise, consider paying for a faster Internet connection or moving to another ISP that offers more speed at the same price.

2. Get a new headset.

Consider upgrading your headset if you use a soft phone to make calls. The thin cable connecting the headset to your computer doesn't last long, and there are many quality headsets in the market. Also consider ergonomics and comfort, especially if you’ll be using the headset for extended periods.

4. Improve your WiFi signal.

You can improve your VoIP QoS—if you’re on a WiFi connection—by boosting your WiFi signal. There are several antennas on the market designed specifically to extend range and improve signal. Alternately, consider upgrading your router and access card to the latest WiFi standard, 802.11n. It’s faster, and prone to less interference.

5. Drop WiFi, use Ethernet.

WiFi networks can be spotty, especially in environments surrounded by concrete walls or floors. Electrical devices and also add interference. Try a wired Ethernet connection if you haven’t been able to improve your WiFi signal. A wired connection greatly reduces interference, and handles data intensive calls better.

6. Pause any large downloads while on a call.

You may experience delays or dropped calls if you’re downloading large files over the same network you use to make calls. Your Internet connection may not have enough bandwidth to handle the increased load. Pause any non-essential downloads before making a call.

7. Design your network for voice and use QoS-enabled network equipment.

There are special considerations when designing your network to support voice traffic. QoS-enabled routers prioritize voice traffic over lower priority network traffic, such as large downloads. QoS-enabled switches perform the same function in the switch. Reference the following document for information on network design and recommended routers and switches:

• Network design
• Recommended routers
• Recommended switches

Verify your speed at www.voipreview.org/voipspeedtester.aspx. Once you’re there, click Start Test:

This site tests your broadband connection and displays the results. In the summary tab, if all the lights display green, your broadband connection is adequate for RingCentral DigitalLine usage. If any of the lights display red, your broadband may not be sufficient to maintain quality Internet calls. If that’s the case, contact your ISP to see if they can improve your connection.

The "MOS" score posted at the bottom provides an overall quality index:

5=Excellent, 4=Good, 3=Fair, 2=Poor, 1=Bad

Most routers work with IP phones and devices automatically. If you’re using a Linksys WRT-series router, you may need to enable port triggering:

Use these settings to enable port triggering for IP Phones or ATAs: 5060-5090, type: UDP 16384-16482, type: UDP Use these settings to enable port triggering for the Call Controller with SoftPhone:
5060-5090, type: UDP
8000-8200, type: UDP

When using multiple Call Controllers on a single network, we also recommend changing the local port number from within each Call Controller.

On your Call Controller:

• Go to Menu > Options. Expand the Account Information tab on the left, then click Connection. • Change the Local Port value from 5060 to another available port between 5060-5090. The port number you select should not be used by any other Call Controller.