Balancing bandwidth with the selection of VoIP codecs doesn’t come up every day. So, let’s think about it. For connoisseurs of music and movies, the past 15 years have marked the opening chapter of a digital golden age. Media consumers now have the power to download over 10,000 songs to their phone or stream films to their TV. But how have we managed to fit our music libraries into our pockets? Two of the key elements responsible for these great leaps forward are bandwidth and codecs.
For businesses making the switch to SIP, these terms shouldn’t be shrugged off as techspeak or telecom-babble. It’s important to understand how the performance of a SIP provider and your communication infrastructure is directly related to the codecs they use and the bandwidth you have. We’ll help explain it to you, while also showing you roughly how much bandwidth you need to get started with SIP.
In the pre-VoIP days, when telecommunications relied on physical trunks to deliver your calls to the outside world, you were limited by the number of circuits in that trunk. If you had a T1 line in your office, for instance, one trunk could support up to 24 simultaneous phone calls. For a call center operation with 20 agents that might suffice; but if you added 10 more agents, you would need to install a new trunk. And it was neither cheap nor easy.
Within the world of SIP, however, the number of lines or circuits do not constrain you. Instead, bandwidth constrains you. If you have five people in your household all attempt watching cat videos on YouTube at the same time, they may run into playback issues if you don’t have sufficient bandwidth to support these streams. The same applies in SIP.
Calculating the Need for Speed
To determine the bandwidth in your office, you can go to sites like http://www.speedtest.net to measure the current speed of your network. When you conduct the test, you will see results for your Download speeds and your Upload speeds. Download speeds are typically much faster than upload speeds. However, since SIP calls require two-way transmission of data, take the slower of the two speeds for this calculation. (Note that talking to your IT manager is also key. There may be Quality of Service measures in place that gives priority to certain types of data over others.)
As an example, let’s say your download time is 45.50 Mbps (megabytes per second) and your upload time is 8.20 Mbps. We would use the upload figure in this case. Since many SIP providers lay out their requirements in Kbps (kilobytes per second), move the decimal to the right by three places. This means your bandwidth is 8200 Kbps.
So, you may be asking: Is that enough for a SIP provider? To find out, we must first talk about codecs.
The term codec is a perfect example of what it does. It stands for coder-decoder, and neatly compresses those 13 letters into 5. There are two classes of codecs, lossless and lossy:
- Lossless, such as those used in ZIP files, where all the information is compressed into a small package, that can be uncompressed without losing any of the original data.
- Lossy, such as MP3, which maximizes compression to reduce the file size while downgrading quality in the process.
A Beatles fan, for instance, would need 8 CDs to capture all 211 songs in their catalog in lossless form. However, by converting them to MP3, they only need about 1 gigabyte of space, leaving room for thousands more. Many music listeners happily sacrifice a minor decrease in quality for the increased convenience.
Voice codecs are mandatory in VoIP because they encode the words of a caller into digital packets that can then be transmitted across the internet. When the data stream arrives at its destination, the VoIP codecs help decode those packets back into sound. Each codec impacts the audio characteristics in a specific way, and each has unique bandwidth requirements to operate efficiently.
Here is a quick breakdown of some of the standard VoIP codecs in SIP and VoIP, including the bandwidth typically needed:
|Codec Name||Bitrate per second||Bandwidth (average)||Description|
|G.729||8 Kbps||24 Kbps||Uses an algorithm for intense compression, helpful in low-bandwidth situations|
|G.711||64 Kbps||80 Kbps||High-quality codec which offers lossless data compression to reduce bandwidth needs; can also be used for faxing|
|G.722||48-64 Kbps||80 Kbps||Offers higher speech quality, but consumes more bandwidth|
|G.726||16-40 Kbps||56 Kbps||Typically used for international trunks|
|G.728||16 Kbps||32 Kbps||Offers toll voice quality for lower bandwidth|
|iLBC||15 Kbps||32 Kbps||Uses a method to handle packet loss which prevents errors from compounding during a call|
A call center operator might be drawn to providers using lower bitrate VoIP codecs since they allow more calls. However, there is often a trade-off in call quality.
Back of the Napkin Estimation
The table above shows the bandwidth needs of each codec vary and aren’t directly proportional to their bitrate. For our example, we will use the G.711 codec, one of the most advanced and popular within the SIP world. It also requires a hefty amount of bandwidth.
Now that we have our bandwidth (9200 Kbps) and the requirements of the codec we’re using (80 Kbps), we need to consider how many simultaneous calls you must support during peak usage. Here, we will imagine a 60-agent call center, where peak usage is 55 calls.
We would multiply 55 by 80 Kbps, which gives us 4,400 Kbps during the busiest time of day. Our 8200 Kbps capacity would be almost double what we would need to support the needs of our call center. However, if the size of our hypothetical call center doubled, we would be wise to secure more bandwidth.
Making the Right Call
Moving to SIP has become increasingly easy, but establishing that your internet connection is robust enough to maintain high-quality voice traffic during the busiest periods is imperative. Learn more to see if you have the right infrastructure in place to support the move to SIP!