VoIP is fast becoming the number one telecommunication option for both businesses and individuals. Technical advances over the past decade mean that call quality is much better than it used to be back in the days of dial-up Internet (using PSTN). However, poor call quality is still one of the most complained about issues with VoIP.
So what are the main reasons for poor quality still persisting? How can these problems be overcome to enable higher quality calls? And what are the other common problems and associated solutions?
Reasons for poor VoIP call quality
The main reason for poor call quality is the underlying internet connection.
It is often the case that smaller businesses with just a few phone lines and a basic package with their ISP attempt to implement VoIP to save money, without really thinking about the impact it will have on their bandwidth.
Because the speed of your internet connection and the available bandwidth have a significant impact on VoIP call quality, it is vital to use a business-class Internet service when implementing VoIP. Opting for fibre optic services will also enhance reliability and speed.
Some call quality issues are down to the fact that VoIP calls are often sharing network resources with a provider’s other internet users. The audio data is essentially having to fight for priority against all other types of data on a public network. One way around this issue is to choose a provider that offers Internet connectivity and Hosted PBX on the same network.
With these providers, because the hosted PBX infrastructure is on the same network, the voice packets don’t have to travel long distances. Additionally, voice packets can be prioritised over any other data transmissions so the VoIP calls don’t have to fight for bandwidth needed for other tasks, for example downloading files, sending/receiving emails with attachments, or participating in webinars.
Another, lesser known alternative way to boost call quality for larger businesses (with 30+ users at a particular site), is to opt for a dedicated circuit. This is essentially a circuit, typically in a T1, fibre or Ethernet over Copper, which is used to connect on-site handsets directly to a hosted VoIP provider’s off-site server, completely bypassing the public Internet (and thus not having to compete for bandwidth).
This gives complete control over latency, packet loss and jitter meaning that the provider will be able to guarantee call quality. Although having a dedicated VoIP circuit sounds expensive, in reality, it will probably cost no more than using VoIP over a third-party ISP. This is because the existing internet connection will be replaced with the new circuit and the client will still be paying for the same things (one dedicated fibre circuit and the Hosted VoIP service).
Other common VoIP and voice termination problems
VoIP termination is the act of routing calls across the internet until received by the intended recipient’s local network which may or may not be on the PSTN.
Frequently dropped calls
This can be extremely frustrating and can also be damaging to an organisation’s reputation and/or customer satisfaction ratings. Three of the most common reasons for dropped calls are poor QoS implementation, issues with internet bandwidth limits, and incorrect equipment configuration or faulty equipment. To combat these issues, QoS should be carefully planned so that packet traffic for voice is prioritised and not delayed due to interference from other traffic. Bandwidth issues can be dealt with by opting for suitable providers and packages or running VoIP traffic on a dedicated network. Correctly configuring equipment, ensuring that any outdated equipment is replaced and upgrades are implemented will also help.
Crackly-sounding calls can be a result of jitter. Jitter is defined as a variation in the delay of received packets. VoIP-based phone services transmit audio as ‘packets’ over the internet. These are sent in an evenly-spaced stream but in some cases they may not be received in the correct order. The result is significantly compromised call quality.
Jitter can be caused by network congestion, improper traffic prioritisation, or configuration errors. Solving jitter may require increasing bandwidth, changing prioritisation, or resolving minor hardware incompatibilities. Moving to a single provider for VoIP and internet may also help reduce jitter.
This is the time delay between the caller speaking and the recipient hearing. Slow network links can cause this and it is usually measured in milliseconds. A latency of below 50ms is recommended and shouldn’t have a negative impact on call quality. The most effective ways of dealing with/reducing latency include policy-based network management, packet categorisation and upgrading hardware.
Whether you already have VoIP or are contemplating switching from analogue/ PRI service, why not take a look around our website to learn more about the products and services that IDT offers.