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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.
Within the world of SIP, however, the number of lines or circuits does 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 to SIP.
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.
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.
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 |
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.