Network Measurement Utilities: PING, Latency, Jitter, and Packet Loss
Network Measurement Utilities: PING, Latency, Jitter, and Packet Loss
VoIP services offered by Vintalk are both robust and cost-effective platforms for delivering communications to business; however, much of what makes our services so great for customers is due to the nature of the service operating over the public internet. The terms and concepts defined in this article are important components in helping assess a customers experience with our services.
(If you're looking for an easier measurement of voice quality see our article on Mean Opinion Scoring)
There are four terms which we use when measuring a customer network connection quality, which is in turn how we determine a customers voice quality potential:
1. PING
PING is a standardized network utility used to test the reachability of a device on a customer network. It is also used to measure the round-trip time, typically in milliseconds (ms), of messages sent from an originating device to a destination device. PING takes advantage of the Echo Reply function built into the Internet Control Message Protocol (ICMP). As a part of normal troubleshooting, a customer may also be asked to allow PING requests (a.k.a. ICMP) through his firewall so that we can monitor customer connectivity and PING times.
For more information, please see our documentation source: ICMP
2. Latency
Latency is the measurement of time that messages in a computer network take to arrive to a particular destination. Latency is measured either one-way (time from source to destination), or more commonly, as the Round-Trip Time (RTT) of messages and their corresponding replies from source to destination plus destination back to source. Round-trip latency is very critical to Vintalk, and VoIP in general, because it is used to determine the amount of time it will take to relay voice and call control data (aka SIP and RTP) to customer locations. Latency is almost always measured using PING and traceroute utilities including the basic utilities included with most VoIP phones, routers, and computers. For high quality Voice services it is recommended that customer network RTT Latency (PING time) average between 50-95ms during peak and off-peak usage hours.
For more information, please see our documentation source: Latency
3. Jitter
Jitter is the variability, over time, of latency measurements between a source and destination network device. An important consideration is that Jitter is a somewhat imprecise term since it relies on determining a mean latency in any one situation, which can change based on a wide variety of variables. We recommend identifying Jitter by working with our support representatives and allowing time for network diagnostics to determine a mean RTT while also taking network routes into consideration.
For more information, please see our documentation source: Jitter
4. Packet Loss
Messages passed between computers in a network environment are commonly referred to as "packets." Packet Loss (PL) is most commonly formed as a percentage measured variable number (i.e. 20%), and is typically calculated using PING utilities to measure the percentage of replies sent to requests. For the most popular protocols which run services like email and webpages, when replies are not received, the service initiating the request will resend the packet; however, for audio/video applications which are streamed in real-time (such as VoIP) resending packets increases overhead and is not practical since humans cannot interpret audio and video out-of-order the same way as a web browser or email client. Customer networks experiencing PL of over 2% will experience degraded call quality and a number of other VoIP service issues.
For more information, please see our documentation source: Packet Loss