Considerations Needed for SIP Trunking

To prepare your company for you need to assess the usage rates of your business communications. Specifically, you should consider how many people are on the phone at the same time during your busiest hours. The answer to this question will determine how many channels you will need. Remember, SIP trunks allow for quick and easy scaling, so you may add or remove channels as needed, if you under or over estimate. You can also decide which method to purchase SIP services. Some organizations may benefit purchasing a set number of trunks while others would benefit from a set number of minutes.

Network considerations that must be examined include total available bandwidth, Quality of Service (QoS), and firewalls. Upgrading your Internet connection may be necessary to ensure sufficient bandwidth to carry UC on top of typical Internet usage for your company.

Use the following simple equation to determine the necessary bandwidth to support your calls:

(number of concurrent calls at your company’s peak) x 85kilobits per second = bandwidth in Megabits per second (Mbps) needed for calls

Equally important to bandwidth is QoS. QoS prioritizes your voice traffic and ensures that your phone calls are going to get the bandwidth needed, regardless of what else is happening on the network. The vast majority of business grade network routers will provide QoS for your network.

A firewall is critical to maintain security both within a LAN and Wide Area Network (WAN). Though firewalls are a critical component to any business network they must be configured correctly to work well with SIP trunking.

For safety, it is essential to add Enhanced 911 (E911). E911 is a feature of the 911 emergency-calling systems that places VoIP emergency callers with the appropriate resources by associating a physical address with the calling party’s telephone number.

Potential Network Issues

On occasion, a PBX with SIP Trunks may experience potential network issues such as Jitter, Latency, and Packet Loss. Jitter can occur when voice packets arrive with varying delays. This is typically caused with changes in network traffic. Latency (delay) doesn’t typically cause audio quality issues, but when the latency is over 150ms, the delay is noticeable to your users. Packet loss, the failure of one or more transmitted packets to arrive at their destination, can cause noticeable effects in all types of digital communications.

These infrequent network issues can usually be fixed by QoS, reducing the amount of traffic your network equipment is handling, and simple network tuning.