VoIP is critical to the way the world works today. It's important to understand how to measure it for your business.
MCS provides key metrics such as jitter, packet loss, the order of packets, and the MOS score. These metrics give an overall view of how VoIP is performing for a specific connection.
It's also important to consider bandwidth when testing VoIP readiness.
Jitter (average) is one of the most important metrics. Jitter defines the amount of time between the packets traveling in the same direction on the network. If the time between packets is too large then a VoIP call will be negatively affected. The negative effects include people talking over each other or distorted sounds.
Our recommended jitter thresholds:
|Good||<= 1 ms|
|Okay||1.1 - 5 ms|
|Bad||> 5 ms|
Most VoIP tests only measure just average jitter. However, that overlooks the importance of max jitter.
The average jitter measurement will, of course, rise as max jitter becomes more prevalent but it's an important metric in its own right as it affects the jitter buffer.
The jitter buffer is a live cache of the conversation so if one party has brief connection issues call quality is maintained as the cache (buffer) can be used. However, too many of these events can cause the cache to run out. Once the jitter buffer runs out it is hard to recover and all connection jitter issues will likely disrupt the conversation audio.
There can be more than one spike of max jitter within a test. Anything more than a couple at 50 ms or higher puts the buffer at risk.
Our recommended max jitter thresholds:
|Max jitter (up/down)|
|Good||<= 10 ms|
|Okay||11 - 49 ms|
|Bad||> 50 ms|
VoIP packets are loss tolerant, meaning any lost packet cannot be recovered. This can have a detrimental effect on the quality of a VoIP call. After all, voice packets contain the conversation. If they are lost then the conversation won't make sense.
A small amount of packet loss can be managed but it doesn't take much (>3%) for it to become a bigger problem.
Our recommended packet loss thresholds:
|Packet Loss (up/down)|
|Okay||1.1 - 3%|
Packet order is a measure in percentage of how many packets arrived in order.
Packets do not necessarily take the same route to reach their destination. This results in packets arriving out of order which causes other packets to be delayed or even, in very bad cases, discarded. Delayed or discarded packets cause a quality problem for VoIP calls.
Our recommended packet order thresholds:
|Packet Order (up/down)|
|Okay||95 - 99.9%|
Packet discards is a measure of packets that arrive too late to be used by the application, which in this case is a VoIP call.
There is a time window when packets can be used after which it is too late and the packet has to be intentionally discarded when it arrives. A bit like missing a connecting flight because the first flight was delayed and arrived after the second flight had taken off.
Our recommended packet discard thresholds:
|Packet Discards (up/down)|
|Okay||0.1 - 2%|
MOS is quite subjective, as it originated from phone companies and used human input from related quality tests. MOS has since been adopted as a measure of VoIP quality.
Our recommended MOS thresholds:
|Packet Discards (up/down)|
|Okay||3.6 - 3.9|
SIP ALG stands for Application Layer Gateway and is common in many commercial routers. Its purpose is to prevent some of the problems caused by router firewalls by inspecting VoIP traffic (packets) and if necessary modifying it. Many routers have SIP ALG turned on by default.
ALG redirection should not be detectable. The SIP ALG test measures and reports the SIP REGISTER, INVITE, and BYE protocol performance while at the same time validating if an ALG is incorrectly allowing internal changes to broadcast back to the test server, i.e. detectable.
A result of Y indicates detectable, which would be a failure. A result of N indicates that ALG was not detectable. This means ALG is off or it is on but not detectable, which is a pass.
Our recommended SIP ALG thresholds:
It's important to reiterate that the thresholds above are guidelines designed to assess a connection for good VoIP quality. Ultimately the thresholds used are a business decision. For example, it may be possible to allow for more average jitter and still maintain calls to a satisfactory level.
Another to consider as part of an overall VoIP qualification strategy is bandwidth. They are quite important as users are rarely just using VoIP. They are browsing the Internet, running web applications, remote desktops, streaming, and much more. All of this adds to the stress of a connection and can cause VoIP issues. Having enough Capacity is key to being able to manage multiple data requirements.
The MCS solution allows you to easily deploy a VoIP/Video assessment portal for your own business. A powerful API allows for endless scope in terms of how the portal is presented to your end-users. Data can be easily segmented and the highly customizable reporting system makes analyzing results easily.
Testing your connection will uncover the true user experience.
Browser-based assessment testing allows you to present a customized portal to your end-users, which allows them to assess their home or work connection.
Satellite technology. Utilize our powerful MCS Satellites (software and hardware) to create testing points across your network. Continuous testing can then be set up between any and all end-points.