"SIP Trunking" is a marketing term for native Voice over Internet Protocol (VoIP) communication services offered by carriers. SIP (Session Initiation Protocol) Trunking service provides communication of VoIP phone calls to an organization's SIP-based PBX (Private Branch Exchange) over an IP network with Quality of Service (QoS) mechanisms to guarantee transmission characteristics (packet loss, delay and jitter) suitable for voice. A SIP Trunking service also includes gateway service to connect VoIP calls to the legacy Public Switched Telephone Network (PSTN). This simplifies organizations' telecom infrastructure and saves money by sharing the carrier access circuit for voice, data and Internet traffic, and removing the need for Primary Rate Interface (PRI) connections.
The steps and principles involved in originating VoIP telephone calls are similar to traditional digital telephony and involve signaling, channel setup, digitization of the analog voice signals, and encoding. Instead of being transmitted over a circuit-switched (i.e. telephone) network; however, the digital information is packetized, and transmission occurs as IP packets over a packet-switched (i.e. data) network.
Key Features of SIP Trunks from Network Communications
- Support G711 and G729 Codecs
- Carried over our very own High Speed Fiber optic network
- Delivered on a 10/100/1000 Ethernet port on our own equipment that provides QoS for all inbound and outbound calls to your SIP-based PBX (Private Branch Exchange)
- DIDs (Direct Inward Dialing) can also be used on our SIP trunks to reach individual handsets on the PBX