SIP trunking is a Voice over Internet Protocol based on the Session Initiation Protocol by which Internet telephony service providers deliver telephone services to customers equipped with SIP trunk-compatible phone systems.
Unlike traditional telephony, where bundles of physical wires are delivered from the service provider to a business, a SIP trunk allows a company to replace traditional fixed telephone lines with connectivity to the public switched telephone network via a SIP trunking service provider on an IP network like the Internet.
Why Do I Need It?
SIP trunks can replace (or augment) your traditional telephone lines. For example, you may wish to use SIP trunking for disaster recovery, cloud services such as call recording, and enterprise call centre deployment, or to provide overflow services during peak call volumes.
With SIP, your technology can grow with your business. Call capacity can be increased on demand to support unlimited concurrent calls, while ensuring you never pay for more than you need.
One of the benefits of SIP trunking is that it substantially improves business flexibility. Because SIP trunking is not tied to specific locations, it allows you to maintain a virtual presence anywhere your SIP trunk provider offers service. SIP trunks may be used to consolidate all of your voice switches into one or two physical locations so you don’t need to rent voice circuits in every city where you have a presence.
Maximize Unified Communications Performance:
With a SIP infrastructure in place, an enterprise can simultaneously deploy UC apps like presence, instant messaging, video conferencing, and more, maximizing UC capabilities.
Converging your voice and data onto one network enables you to cancel subscriptions to multiple voice circuits. This results in significant savings in hardware costs and long distance charges.
SIP trunking enables you to centralize business continuity and disaster recovery.
Why Choose SIP Trunks over Traditional Telephone Lines?
The possibility for a rapid return on investment is a key driver of SIP trunk deployments. Adopters of SIP trunking have reported up to 40% savings on telecommunications.
SIP trunking delivers the following benefits:
¨ Replaces or augments costly telephone lines and PRIs (Primary Rate Interfaces) services.
¨ Removes the need to invest in additional PSTN gateway capacity as you grow.
¨ Reduces capital expenditures: Edge devices offer a lower investment path in adding new lines as they are typically less expensive per line than the corresponding PSTN gateway.
¨ Optimizes bandwidth utilization by delivering both data and voice via the same connection
¨ Maximizes flexibility in dimensioning and usage of lines as you avoid having to buy capacity in chunks of 23 (T1) or 30 (E1) lines
¨ Long Distance rates are generally lower on SIP trunks
¨ Enables redundancy with multiple service providers and links.
¨ For multi-site organizations:
- Provides centralized trunking — no need to have costly telephone lines or PRIs at all sites.
For multi-site organizations:
Provides the ability to deploy one phone system across all the sites, enabling station to station dialing amongst the sites, and centralized or distributed trunking.
How Do I Deploy It?
There are some details necessary to address in order to successfully deploy SIP trunks:
¨ A SIP Trunk compatible PBX equipped to accommodate SIP trunks
¨ Internet telephony or SIP trunking service provider (ITSP)
¨ Always ensure that the SIP service provider you select is certified to work with your SIP trunk-compatible PBX
¨ Firewall or Session Border Controller (provides enhanced security)
¨ Extensive testing ought to be performed (with your PBX) prior to going live on the proposed SIP trunks (test all functionality, such as add-on conferencing, “0” Out, voice mailbox messages, multiple transfers of single call, ACD groups etc. to ensure full duplex audio.)
¨ In a multi-site deployment with centralized SIP trunks (e.g. in a data centre, or Head Office/HQ) ensure that your available WAN bandwidth between sites is adequate for the expected voice traffic.