SIP Trunking is the de facto standard for VoIP applications. A SIP Trunk is primarily a concurrent call that is routed over the IP backbone of a carrier using VoIP technology. SIP Trunks are used in conjunction with an IP-PBX and are thought of as replacements for traditional PRI or analog circuits. The popularity of SIP Trunks is due primarily to the cost savings of SIP along with the increased reliability as backed by the SLA’s of SIP Trunk Providers.
SIP Trunking is simply a single conduit pipeline for multimedia elements (voice, video and data). SIP Trunking reduces or eliminates the need for PSTN media gateways as well as reduce or eliminate the need for narrow-band voice circuits. SIP Trunking provides a smart and cost effective solution to customers by eliminating the need to purchase additional equipment, such as managed media gateway devices to interface between IP voice to the PSTN.
SIP Trunking provides the following benefits:
- Works with any SIP Supported device
- Additional cost savings may be realized through converged access
- Eliminate the need to purchase and manage traditional TDM-based voice circuits with limited scalability