How Your Internet Connection Affects Call Quality
To ensure the best sound quality during calls, Jive uses uncompressed audio codecs (G.722–HD Voice and G.711-ulaw) delivered through Real-time Transport Protocol (RTP); which allows the audio to stream in real-time. As a result, your internet connection is the most important component to call quality.
The following are important internet requirements that can affect the quality of your calls.
How to test your connection: Jive can do this for you! You can use Jive View™ to test if your system will work well prior to implementation. Jive View is a small application that runs in the background which provides valuable networking data in regards to using VoIP. You can learn more about Jive View and download the application for free here.
For those who like a more hands-on approach; you can test for these metrics by using sites like speedtest.net. Test to a central location (like Chicago or Dallas-Fort Worth) to get an idea of what your connection looks like during peak hours.
Requirement: ≈82.5kbps up/down for each active call
The rate at which data is carried over the internet from one point to another in a given time period (usually a second). Each active call uses approximately 82.5kbps of bandwidth for uploading and downloading call data.
What the other person hears. Audio is uploaded in real-time from your phone to Jive Cloud using your internet connection’s upload speed.
Note: The upload connection is more prone to call quality issues because upload bandwidth is less available on most internet connection types.
What you hear. Audio is downloaded in real-time from Jive Cloud to your phone using your internet connection’s download speed.
While the bandwidth requirement is minimal, you have to take into account how many calls are taking place simultaneously at any given time to ensure call quality is maintained. For example, if there were 10 phone calls happening at the same time, your bandwidth requirement would be 825kbps up/down. Not only that, consider what other types of internet activity (file transfers, video streaming, internet browsing, etc.) are taking place on every computer or mobile device within your network.
Because each device is fighting for bandwidth, Quality of Service (QoS) can be implemented to dedicate bandwidth to phones during periods when internet usage may be saturating your connection.
The average time it takes packets (audio) to travel from Point A (phone) to Point B (Jive Cloud) and back. Many people, including internet service providers (ISP), only consider bandwidth when evaluating internet speeds. However, that is only half the picture. Bandwidth only shows how much internet traffic that can be pushed through; where latency shows how fast that traffic arrives at its destination.
Think about driving on the freeway. Bandwidth represents the number of lanes that are available—if you have more lanes, more traffic can be pushed through and the likelihood of a traffic jam is reduced. Latency represents how fast you drive—it doesn’t matter how many lanes there are if other things are slowing you down (inclement weather, gravel, potholes, etc.).
To avoid issues with call delay and call quality, it is recommended that latency does not exceed 100ms on average (use a ping test to analyze latency).
Jitter (Packet Delay)
The change in the amount of time it takes for one packet (audio) to move from Point A (phone) to Point B (Jive Cloud). When you are checking your email or casually browsing the web, it doesn’t really matter when packets arrive or if they arrive in order—in most cases you will never notice. But when you are streaming media, like a phone call, this packet precision becomes extremely important. If there is excessive jitter, packets will be dropped and call quality will be affected.
Jitter close to 0ms is ideal, but it should not exceed 10ms.
The percentage of packets (audio) lost while traveling from Point A (phone) to Point B (Jive Cloud). If packets are lost, audio will be dropped and the sound quality will be compromised.