Businesses are turning to VoIP for substantial cost savings and greater control over voice communications. But adding IP-based calling to an existing network can raise concerns about how much bandwidth it’s going to require and whether the infrastructure will be able to cope.
With data files, audio, video and more all fighting for bandwidth, there are bound to be some conflicts. You can use QoS settings to manage bandwidth allocation, but sometimes optimisation isn’t sufficient and you need to add more bandwidth.
Codecs used by a typical international VoIP wholesale provider
The first thing to look at when you are considering a switch to a reputable VoIP provider such as IDT is how the technology uses bandwidth. Essentially VoIP reformats voice traffic into data packets. These then travel around the web in the same way as any other files before getting converted back to audio at the destination.
Each packet also has overhead information containing details about its origin and destination that allows it to be correctly routed. This is known as a wrapper; there can be several layers of wrapper information and each adds a small amount – although only a few bytes – to the packet’s size, thus needing a little more bandwidth.
VoIP systems use codecs to convert voice into data. There are different codecs and they offer a trade-off between quality and bandwidth. The higher the call quality you want, the more bandwidth will be required.
The most commonly used VoIP codecs are G.279 and G.711. The G.279 codec uses a compression algorithm to reduce the size of data packets but this comes at the price of slightly lower sound quality. G.711, on the other hand, has no compression but this means it uses more bandwidth. An hour of uncompressed traffic is likely to be around 85 MB but this becomes 35 MB with G.279 compression. That doesn’t sound like a massive difference in today’s world where we take terabyte storage for granted, but if you multiply it by hundreds or thousands of calls each day it adds up to a significant burden.
Understanding packet sizes
So, how big is a VoIP data packet? This is a hard question to answer because there are several variables involved. We’ve seen that each packet contains wrapper information as well as the call data.
The wrapper size is generally constant, 20 bytes for the IP header, eight for the UDP (User Datagram Protocol) header and a minimum of 12 for the RTP (Real-time Transport Protocol) header. To this, of course, is added the data itself which can vary in size.
We’ve seen that packet size can vary due to the codec used and whether there is compression. A larger packet size means fewer wrappers, but again there’s a trade-off in that if a packet gets lost you can lose a significant chunk of the call. Smaller packets mean more wrappers but make the system more tolerant of dropped packets without significant loss of quality.
