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SIP Trunking is simply a protocol for voice, video and data. It's a lot like PRI, except that it operates over IP and give you back your bandwidth for other uses when you aren't using it for voice.
SIP Trunking can be directly accessed through IP PBX's or by simply using PSTN media gateways with existing PBXs and KSUs from Nortel, Avaya, Cisco, NEC, Tadiran, Toshiba, etc.
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; additionally, SIP Trunking provides the following benefits: - Works with any SIP Supported device
- No need to invest in costly TDM gateway equipment infrastructure or desktop equipment
- No need for equipment changeover or disruption to services.
- Additional cost savings may be realized through converged access.
- Eliminate the need to purchase and manage traditional TDM-based voice circuits with limited scalability.
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 SLAs of SIP Trunk Providers. SIP Trunking Platform
Technical Overview |
Level 3 Network Map 
