Voice over IP (VoIP) bandwidth requirements vary depending on the codec used. When you begin to calculate how much bandwidth you’re going to need for your VoIP solution, you cannot expect that every line will be used concurrently and packets will constantly be transmitted and received. Typical voice conversations have moments of brief pauses and silence where no packets are transmitted. Here are tips and ideas to help you calculate your VoIP bandwidth requirements.
If you’re using a dedicated data line or Session Initiation Protocol (SIP) trunking provider, you must make sure that your bandwidth is adequate for your VoIP solution. Typically, an SIP trunking service provider calculates the bandwidth you need if it is being used through a dedicated data line, depending on the number of concurrent calls and the call quality each connection requires.
SIP trunking has been an option for US businesses for nearly a decade and continues to grow. One of the main questions companies have is if their current private branch exchange (PBX) supports SIP trunking, which should be easy to find out by doing some basic research on the PBX vendor.
As mentioned, VoIP call quality is only as good as the codec it resides on. The codec converts voice into compressed digital data to transmit it over the broadband connection, which is then converted on the other end for the listener. This process goes on throughout the length of a VoIP call.
The codecs typically used are G.711, G.722, G.723, G.726, and G.729, all of which feature digitized voice signals and require various rates of bandwidth, from 8 Kbps to 64 Kbps, respectively. The most commonly used codec is G7.11, which requires 64 Kbps of bandwidth and provides good voice quality to an SIP trunk that has a dedicated line. When tallying the bandwidth, don’t forget the 23 Kbps of IP overhead required per call, bringing the total to the 87 Kbps required per VoIP call using G7.11.
Each of the commonly used codecs is recommended for different situations. For example, G.729’s 8:1 compression ratio offers great quality for mobile call quality. If an SIP trunk joins the provider online, the call quality is drastically affected if there is a lot of normal Internet traffic that drops or delays calls, causing patchy voice or static, which is why dedicated IP connections for VoIP traffic are the best option, if available.
Latency is the biggest issue for VoIP. It is measured in milliseconds and involves the round-trip delay between the transmitter and receiver of a call. The ideal latency for a call is in the 250–300-ms range both ways and about 100–150-ms one way. The greater the distance between the sender and receiver, the greater the latency. You can test your latency by sending a ping to your SIP provider’s trunk end point, which will show you if you have a latency issue. Knowing how to calculate your bandwidth requirements before adding an SIP trunk service will help you avoid issues such as latency by simple tests and understanding of codecs.
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