SIP Trunking

Primarily, a SIP Trunk is a secure communication tunnel allowing concurrent calls to be routed over the IP backbone of a carrier using IP technology.

SIP Trunks are used in conjunction with an IP-PBX (Private Branch Exchange) to replace traditional PRI or analog circuits and are fast becoming the standard technology that businesses use to make and receive telephone calls at a lower cost and with more features.

SIP Trunking can also be used with traditional PBX systems. Connections can be made to a PBX via analog Business Lines or PRI Services. SIP Trunking can work for Legacy PBX Systems by using an IAD (Integrated Access Device) that allows SIP Trunks to seamlessly connect with the system.

By serving as a converter between a legacy phone system and a company’s Internet connection, a SIP-Trunking device allows the data network to carry voice traffic. SIP Trunking can blend connections for both data and voice into a single line. The popularity of SIP Trunks is due to cost savings, increased reliability, added features and flexibility.

We Can Deliver SIP Trunking Connections In 3 Ways

Converged Voice and Data in One Pipe

This is for customers, who want to use NET PLUS GROUP for Both the Voice and Data services. This is the most cost effective way to implement SIP Trunking into your Business.

A dedicated Connection for VOICE ONLY (without Touching the existing Data Network)

This is for Customers who do not want to change or mix their Data Network and Connection with Voice. NET PLUS GROUP Can install a dedicated SIP Trunk connection for VOICE ONLY. This will by passes all existing Data Infrastructure while keeping Voice on a Private Network.

Using your Existing Data Connection for SIP Trunking

This is only recommended for Customer who have existing T1 or Fiber Data connections and have the technical expertise on staff. The Quality of Voice Service will be dependent on your Data Providers Data Connection because we have no control over the last mile.

We provide a variety of Network Protocols – UDP (most commonly used), TCP and TCP/TLS for more secured communications. If required SRTP (Secure RTP – Secure Real-Time Protocol) can also be provided to anyone with Higher Security in mind.

On top of that we provide support for different Audio Codecs such as G722, G711 (uLaw/aLaw or PCMU/A), T.38 (T.38 and T.30 compliant Faxing Protocol) as well as Full-Band Opus (Opus is a modern full-band Codec which is highly adaptable to the latency and Internet bandwidth using 8Khz/16Khz/32Khz and 48Khz bands).

We also provide up to a degree Video Codec such as H.264 and VP8.