A crucial step of planning for a VoIP service is calculating the bandwidth that will be needed.
Bandwidth requirements vary depending on the codec used. Furthermore, a business can’t expect every line will be used concurrently, and packets will be transmitted and received. For instance, normal conversations include a lot of silence, which often means no packets are sent at all.
When using a dedicated data line or Session Initiation Protocol (SIP) trunking provider, it’s important that a business ensures they have adequate bandwidth.
Generally speaking, a SIP trunking service provider will calculate the bandwidth a business needs if it’s being used through a dedicated data line, depending on the number of concurrent calls and the call quality each connection requires.
If a business wants to find out whether their current private branch exchange (PBX) supports SIP trunking, they should do some basic research on the PBX vendor.
Before setting up a VoIP service, a business can test their internet speed using an online tool and should consider their bandwidth allocation.
It’s important to recognise the number of call paths needed as this can help a business determine the amount of bandwidth needed. A business should ask itself whether they will have enough bandwidth when all call paths are being used in addition to regular internet data usage.
Codecs convert voice into compressed digital data to transmit it over the broadband connection. The quality of a VoIP call is dependent on the codec.
Codecs are typically G.711, G.722, G.723, G.726, and G.729. All of these require different rates of bandwidth, from 8 Kbps to 64 Kbps.
The G.711 codec requires 64 Kbps and is the most commonly used by businesses.
This codec will generally provide good voice quality to a SIP trunk that has a dedicated line.
On top of this, 23 Kbps of IP overhead will be required per call, meaning the total bandwidth required for a G.711 codec is 87 Kbps.
Each of the commonly used codecs are recommended for different situations. For example, G.729’s 8:1 compression ratio provides quality mobile calls and may be a consideration for a business with remote workers.
Latency is a significant issue for VoIP. The ideal latency for a call is in the 250-300ms range both ways, and about 100-150ms one way.
The greater the distance between the sender and receiver, the greater the latency.
One way to test latency is by sending a ping to the chosen SIP provider’s trunk end point, as this will show a business if there are any latency issues.