How does network performance impact hosted voice?

Your network performance is crucial to the performance of your hosted voice deployment. Often forgotten about, the state of your network plays a fundamental part in your hosted voice performance.

The last thing any IT Manager wants is for a new hosted voice solution to be implemented – only to be forever resolving quality issues.

When preparing for a hosted voice deployment, guarantee the quality of your voice calls by taking note of these network concerns ahead of time.

Latency

Latency cannot be completely removed. It will always take an amount of time to transfer speech from real time to the other end of the phone. When the round trip latency goes above about 250ms, it becomes noticeable on a phone call. This is usually where you experience people speaking at the same time or waiting for longer than necessary to ensure all speech has made it through. This both unproductive and a poor representation of your business.

Latency is caused by poor packet handling and queuing delays. To remedy this, the network must have sufficient bandwidth and be configured to prioritise voice traffic.

Whilst you cannot control the external environment, your internal network latency should be no more than 150ms for your hosted voice to operate at optimal performance.

It is important to note here that the genuine quality is not impacted, just the delivery time of the speech.

Jitter

Jitter is the variation in the delay of received packets. High jitter means patchy speech and temporary - or ongoing - glitches. VoIP uses algorithms to compensate packets that arrive at high timing variations, but packets still get dropped when they arrive with excessive delay.

Hosted voice solutions employ packet buffers to compensate for instances of jitter. This makes it more difficult to identify genuine problems on calls. High jitter rates mean dropped packets and gaps in speech transmission. Your internal network jitter should be operating at less than 100ms. A higher level of jitter will cause voice transmission delay and sometimes packet loss. Jitter is often caused by network configuration and the lack of prioritisation set for voice traffic.

Packet loss

Packet loss impacts the quality of your voice calls.

Voice traffic has to be delivered in real time. This differs to other internet traffic like email, browser and application activity which is unnoticeable if delayed by milliseconds or even seconds. If there is a second or seconds delay during a phone conversation, it is immediately noticeable and incredibly interruptive.

If packets are late or lost during delivery, speech is delayed or completely inaudible. Packet loss is sometimes caused by either latency or jitter. More often, it is caused by network configuration, lack of bandwidth, or dated equipment not designed to handle voice transactions.

Mean Opinion Score (MOS)

Hosted voice platforms use Mean Opinion Score (MOS) as a standard measure to score conversation quality over any given network. MOS is scored by measuring each of latency, jitter and packet loss wherever a conversation occurs. This could be in your main office, supported by your network. It could be in a user’s home office or even on their mobile.

The common rating scale is the Absolute Category Ranking (ACR).  This breaks down the scoring into the below levels:

5 Excellent

4 Good

3 Fair

2 Poor

1 Bad

With optimal network performance, calls should always be performing between 4 and 5. As part of your hosted voice deployment, you will have access to MOS reporting. This will give you access to the score of every call made over your voice platform. Advanced analytics will also provide you with detail as to where and why any issues occurred, including those directly related to your network.

Resolving network performance

Voice quality issues can be hard to track down, especially when they are intermittent. But, they can have a huge impact on your hosted voice performance and your business. One example is the CEO of an investment finance company who told us they were having real problems with conference calls with high value clients. Due to the poor network performance, the voice stability and quality impacted customer facing interactions, portraying the CEO, and the brand, in a bad light.

To pin down voice quality issues, here are some recommended first steps:

Network Upgrades – When moving to hosted voice, ensure your network equipment is fit for purpose. A top spec voice router and enough bandwidth via your internet service provider are good starting points here.

Network Configuration - Prioritise VoIP traffic to improve both latency and jitter. This is typically achieved by using bandwidth reservation, policy-based network management, Type of Service and Class of Service.

Network Monitoring – Without a clear view of your network, you won’t know where the problems are occurring.

Network Monitoring

The adoption of hosted voice will have implications on your network. If your network can’t support hosted voice performance, this needs to be resolved before you go live with hosted voice.

To ensure your network is operating readily for hosted voice, and to ensure ongoing performance, you must monitor your network and infrastructure appropriately. Monitoring helps to spot basic issues, but also more subtle performance issues.

SAS has one of the best, if not the best monitoring and management platforms in the industry, giving you complete, deep and interconnected information about the network and about how your applications are performing over it. We show you the complete application path, so you can easily spot when and where problems have arisen and do something about it.

To learn more about how SAS enables your network to empower your hosted voice solution, book a call with one of our network specialists.

If you would like to find out more, then you may like to read one of our other blogs or downloads, or request a conversation with one of our consultants. We are always happy to discuss, without obligation.