It happens to every business. A customer tries out your product only to inform you it doesn’t work right and they are no longer interested in your product. Fortunately, you don’t like taking no for an answer easily so you press them to find out why they think it doesn’t work. After a little digging you discover it has nothing to do with your product or service but rather something on their end which you have no control over.
When I worked for a small software company it was inevitable that we got blamed if anything went wrong with the computer where our software was installed. We would spend hours of our time troubleshooting only to discover that either they didn’t install the software properly, had an incorrect router setting preventing links from working, or as in many cases, they had a computer virus.
It’s no different in the telecommunications world. While in software, it was often a virus which was the culprit, in our case it can often be the bandwidth of the client’s internet connection which affects the quality of voice over the internet.
Therefore, we thought it would be a good idea to put out there in this article some details on the basic principles of bandwidth allocation as it relates to SIP trunking and its implications.
One of the preliminary steps which you should consider before pursing SIP trunking is to take analysis of your existing bandwidth availability. To do this, you will need to do the following:
Test your internet speed
Go to www.speakeasy.net and use their speed test tool to determine your internet speed. It will test both download and upload speeds and display them on the screen. You can choose a city near you to test through or one farther away. The farther away the slower the speed.
Consider your bandwidth allocation
If your business already uses the existing internet connection for heavy data use and plans on using the same connection for heavy call volume, you might run into a problem if the connection you have is not broad enough to handle the load.
This doesn’t mean that you will not be able to handle SIP trunking however. There are other options such as getting a dedicated T1 for your voice (or data), and in some circumstances securing an MPLS (Multi-Protocol Label Switching) which might be the best option depending on the needs and requirements of your organization.
Determine how many call paths you would need
This is important because by recognizing the number of call paths needed, you can determine the amount of bandwidth needed. The question becomes will you have enough bandwidth when all call paths are being used in addition to your regular internet data usage?
To help you determine what you need to look for, we have created a graph which shows how much bandwidth in MB you will need for the allotted call paths. If you don’t meet the recommendations and still proceed with a voice over IP solution like SIP you will most likely experience a degradation in call quality as a result of dropped call packets.
The two charts below show the required bandwidth in MG (Megabytes) as well as the number of T1s which would be required to provide the necessary bandwidth.
The first graph is if you are using a G.711 Codec which uses 80-85k of bandwidth per call path. So for one call path you would only need about 80+k to make your calls without any quality issues. The greater the bandwidth allowed such as in the G.711 vs. the G.729, the better your voice quality will sound.
In our graph we broke it down with our bundles for the sake of simplicity and practical application. Starting with our 50 call path bundle, you would need approximately 4MG (50 paths x 80k = 4000k). This would require three T1s (one T1 = 1.5MG). Since you need 4 MG, you would need three T1s to provide adequate bandwidth.
The second graph shares the same scenario, but using the G.729 codec which uses less bandwidth. Remember, the less bandwidth allocated for each path the lesser the quality of voice the user will experience. However, for EtherSpeak, the G.711 is the default used, but if a user has bandwidth constraints, then we will go ahead and change it to the G.729 codec.
If you are able to provide the appropriate internet bandwidth which will cover “both” voice and data, you then should move forward with testing out some SIP trunks. If not, then you have just spared yourself from having to use up your time trying to troubleshoot quality control issues with your carrier when all along it was a result of your own lack of bandwidth.
Bandwidth of course is not the only factor when factoring SIP trunking quality. Issues such as latency, jitter and packet loss can also cause quality issues. These types of issues result in echos or when you find yourself and the other caller consistently interrupting each other.
Without going into further detail (save that for another post), it’s always good to be aware that there are many issues which can affect voice quality over the internet. If you have a strong connection to start with you are at least off to a great start. Make sure when looking for a SIP provider, that they provide you with a trial and are able to discuss these types of issues with you.
If you are ready to look into SIP trunking as an option for your telecommunication needs for your business, feel free to give us a call at (866) 384-3747 or contact us online and we’ll be glad to answer any questions.
Request a free quote and trial here.
Liz
EtherSpeak, Inc.
www.ietherspeak.com





