This article will run through the steps required to diagnose any issues with your Jet apps or your network connectivity.
- How do our calls work?
- Accessing your call quality report
- Understanding your report
- Improving your Network or Router Setup
How do our calls work?
Phone calls made via your Jet Interactive Services are placed over your internet connection using a type of traffic called VoIP, which stands for Voice over Internet Protocol. A VoIP system works by taking your analogue voice signals, converting them into digital signals, and then sending them as data over your internet connection.
This means the most common reason for your Jet Phones or Hardware to not be connecting is due to Network connectivity issues. If your internet connection is not strong enough or not configured to support this type of traffic, you can occasionally experience call quality or app connectivity issues.
Jet provides a range of reports on Device Quality & Network reports that you can use to check the frequency of issues on your services.
There are also some features custom-built into your Jet Phone app to help diagnose and detect these issues, and there are certain router settings and permissions that you can modify yourself, or speak to your technical contact or Internet Service Provider to modify on your behalf.
Accessing your call quality report
On mobile devices running iOS or Android, the Call Statistics report can be accessed through the Advanced Settings menu on the mobile application.
Once you have loaded the report, take a screenshot straight away. This report will be wiped and replaced when the next call is taken on this device.
On the desktop versions of our apps (Mac & Windows), this report is available by using your menu to navigate to Help > Troubleshooting > Audio.
Again, you need to screenshot this report ASAP -- either mid-call or directly after. Subsequent calls will overwrite this with new call information.
Understanding your report
That's a lot of technical jargon in the report, so let's break each term down for you in more detail.
Each of these results has different methods to solve, so please follow the information at the bottom of each section to view the steps to resolve.
Here are the key things to look for in your report:
Codecs are device programs or drivers that encode and decode audio signals to be able to pass the signal through telephone lines or over the internet. For voice calls, they are used to translate the sound of your voice into a digital signal, and then produce it on the other end for the other party to hear.
The most common audio codec used is called G.711 (also called PCMA) or its higher quality brother G722. These codecs are not compressed, meaning they may use up more of your bandwidth connection than other codecs, but this guarantees a higher quality (HD) voice call than most other codecs. When using this codec, the Throughput in your Call Statistics report should sit between 80-95kbps.
Other less common codecs include G.729 (some compression, designed for slower internet connections) or G.723.1 (highly compressed to save bandwidth, but sacrifices call quality).
Your codec will automatically adjust to the best quality available based on your bandwidth. If your codec is not G.711, PCMA or G722, please view our speed guide for the minimum download speeds you should be aiming for to get the best quality call and consider upgrading your network connection speed.
Packets & Packet Loss
When a call is made, the signal gets recorded and then split into thousands of tiny pieces, known as packets. Packets are usually encrypted and are very small in size to provide easier and fast data transfers between endpoints. Packets are all sent individually and reassembled on the other end to complete your audio signal.
For a 5-minute call, this is about 15,000 packets being sent.
When packets get Lost, it means that some of the data went missing along the way and was unable to be delivered. You can see the percentage of Lost data in your Call Statistics report.
Packet loss is usually due to network congestion or insufficient bandwidth. Please view our speed guide for the minimum download speeds you should be aiming for to get the best quality call and consider upgrading your network connection speed or making changes to your bandwidth control to ensure VOIP traffic is prioritised.
Sometimes when passing data from point to point at such high speeds, not all the data will be sent and received at the exact same time. This is called Jitter and refers to the variation in milliseconds between these packet delays. This is the main reason for poor audio quality in a call, usually in the form of choppy sound.
Jitter should be as low as possible, ideally below 5ms. If your Jitter is very high, please view our speed guide for the minimum download speeds you should be aiming for to get the best quality call and consider upgrading your network connection speed or making changes to your bandwidth control to ensure VOIP traffic is prioritised.
Our system also has an adaptive feature called a Jitter Buffer. This feature will identify if the call has impacted quality due to patchy or low quality internet, and the system will actively work to improve the call quality.
This feature can take some time to kick in (usually resolves within 10-15 seconds), and if you are finding your call is crackly for the first few seconds and then improves, this is thanks to the jitter buffer feature. Please follow the links above to check your bandwidth allocation and check your speed guide to improve, and note that poor reception in your area could also be a factor.
Round Trip and Latency
Round Trip is the time in milliseconds that it takes for the sound to travel between the two ends of a call. You may sometimes see this referred to as latency by other service providers, and a high latency is indicative of a delay in transmitting sounds.
The lower the Round Trip or Latency, the better - ideally this should be under 25 milliseconds. Most of the time, latency and extended RTT result from network congestion, which also contributes to jitter (above).
If you have high RTT, please view our speed guide for the minimum download speeds you should be aiming for to get the best quality call and consider upgrading your network connection speed or making changes to your bandwidth control to ensure VOIP traffic is prioritised.
Please note - using our VoIP services when located outside Australia can lead to higher RTT than expected. If the steps above are unsuccessful in improving this, we recommend talking to your ISP for more information on how to improve this issue specifically from your country of origin. You can also consider investing in hardware that will support your unique needs for your location, although Jet cannot provide further guidance here, and we recommend speaking to a network provider or IT support in your location for specialised advice.
Your upload/Download speed (known as bandwidth) is the maximum throughput of your Internet network.
Bandwidth greatly affects all the components of your voice call. Bandwidth congestion is the main cause of packet loss and compromised call quality, as well as "dead" lines and one way audio. In this case, the higher your bandwidth, the better.
If your bandwidth is under 80kbps, please view our speed guide for the minimum download speeds you should be aiming for to get the best quality call and consider upgrading your network connection speed.
For more information on improving your Network setup, please view our detailed information here: Improving your Network or Router Setup