Although most people consider enterprise-grade features and cost-friendliness to be the number one reason that businesses trust VoIP systems more and more, there is one thing that is not mentioned quite as often: crystal clear, uninterrupted voice calls. However, even if the VoIP service does provide HD-clarity voice calls, it is worth nothing if the calls are constantly dropping.
And not being able to properly place and receive calls could lead to serious disappointment in VoIP, even though in many cases the issue and its solution may be right under your nose.
Long story short, basically any situation in which the call is abruptly ended before either party has finished their conversation should be considered a dropped call. However, dropped calls can appear in many different forms including, but not limited to, the following:
As you will see, there are many instances where the symptoms are eerily similar to one other. And that’s why dropped calls can be so deceptive, because not being able to read the signs properly may result in prolonging the discovery of the issue and ultimately leave you with a faulty phone system.
Admittedly, memorizing all these types of dropped calls can be challenging, but it’s still a better option than doing nothing. In fact, it’s an important part in a clever method with which symptoms can be identified in a jiffy, therefore giving way for the best and often long-term solution.
The first step to figuring out which type of dropped calls is poisoning the system is to direct some of the following questions to your colleagues:
Once these are answered you can immediately move on to identifying the issue and finding the right solution.
A fast busy signal usually indicates interference, which is the result of having too many devices – particularly ATAs or routers – hooked up to the same network. To eliminate this problem you may need to either test each device until the faulty one is found or reduce the number of networking devices.
In this case testing the entire network for latency is the right solution. If that latency is more than 150 milliseconds, then your network can’t handle voice traffic properly and ultimately leads to audio issues. To prevent this from happening again try to use a bandwidth saver – like VirtualPBX’s VoIP Clear Fix Service or Vonage‘s built-in solution – or simply opt for a better internet service.
There are various reasons why calls may not be completed. Sometimes it’s as simple as misdialing a phone number, however, more often than not there are instances where the issue is at the call recipient’s end, and so the fix is left to the VoIP service provider.
In this case it’s likely that the service provider is limiting calls to either 120 or 180 minutes. Since the termination of calls after a certain amount of time is a built-in safeguard that prevents users making fraudulent calls and getting large bills – and is therefore not a bad thing at all – the only good recommendation here is to avoid exceeding the maximum call time.
When the end of sentences are simply cut off by the system it means that silence suppression isn’t properly set on the phone. To solve this problem, go to the settings of the phone and adjust silence suppression until all sentences can be heard in their entirety on both ends of the line.
VoIP is a fully internet-based technology, meaning that you’ll need access to an internet connection and devices that should be able to handle not just regular, everyday internet traffic, but additionally cope with incoming and outgoing calls, too. However, in order to learn whether the current network is VoIP-compatible, future customers will need to do a quick check first by doing a so-called ping test.
A ping is a diagnostic tool available for all major OSs that provides data on how stable the communication between a device like a router or even a VoIP phone and another endpoint on the internet is. By running a ping test the computer will highlight how many (voice) packets failed to make it to the destination. If none of these packets are lost during transmission then it’s safe to say your network is ready for accepting and placing VoIP calls without major disruption.
However, the best way to ensure your network is VoIP-ready is if you prioritize internet call traffic over everything else or – and this is the best option – purchase a VoIP router that is configured by default to handle internet telephony.
Performing the ping test is ridiculously easy and it can be done without even accessing an internet browser. On Windows you just need to right click on Start, select Command Prompt, type ‘ping’ and the IP or website address of the future VoIP provider, hit Enter and wait for the results. On a Mac the command can be launched by opening Terminal and entering ‘ping’ with the host’s IP or address; note, however, that the test won’t stop until you press Ctrl + C.
There is also the option to do the whole test with the free online tools from companies like OnSIP and MegaPath. These tests take into consideration various factors like upload and download speed, jitter and latency (often called ping or delay). Taking out the upload and download speeds from the equation, every other value should be as close to zero as possible in order to consider a network ready for VoIP.
The various ping tests we ran to learn about our network’s VoIP capabilities had promising results: all tests measured less than 150 milliseconds of latency and less than 3 seconds of jitter, meaning that the network is ready for placing and receiving VoIP calls, both for voice-only and for video feeds too.
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