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V1 VoIP Explains How SIP Trunking Works

Posted on: August 25, 2017

V1 VoIP explains how does SIP Trunking work with PBX and VoiPResellers and wholesale agents tell V1 VoIP all the time that SIP trunks are in high demand from their clients. A major aspect of VoIP technology is SIP and SIP trunking. If you are interested in offering SIP trunking to your clients it is best to have a complete understanding of what they are and what they do. Here V1 VoIP explains the fundamentals of what SIP trunks are and what their capabilities are.

SIP stands for Session Initiation Protocol, and it’s a type of protocol that initiates a session over the Internet. (Protocol is a set of rules governing the format of messages that are exchanged between computers.) SIP Trunks can save an average small business could easily save 50% or much more on their monthly communications bill because they are more cost effective.

A SIP Trunk is Switched, which means incoming calls from the SIP trunk to the IP-PBX is switched to other devices sitting behind the interfacing CPE proxy device. A SIP trunk offers the IP-PBX switched access to a diverse set of PSTN and/or on-net IP termination points via V1 VoIP’s network side of the SIP trunk.

A SIP trunk establishes a pure IP connection. The VoIP trunk can handle the multi-media options of voice, data, video and mobile connections. The circuit-switched hand-off to the public switch telephone network (PSTN) is not needed, which will lower costs. DID (Direct Inward Dialing) numbers can be provided from virtually any location in North America and a large number of other countries. This permits enterprises to establish a “virtual” presence with incoming calls ringing at a centralized facility. Direct SIP trunking to IP PBXs (private branch exchanges) and soft switches means no additional hardware to achieve end-to-end VoIP.

Traditional voice is attached to a physical location, but the flexibility of V1 VoIP’s SIP allows call routing to any IP address worldwide, including home offices and soft phones. V1 SIP Trunks are able to route calls with any variant of DID or 800 numbers to a specific client on the SIP trunk with multiple ENUM termination. This means that the SIP trunk is not limited to a single 800 or DID number, as it is a switched service that the CPE Proxy will finally route in its domain.

The SIP trunk connects with a SIP device at the customer premise that is a proxy device, with one or more user agents logically attached to it. A border controller/firewall is provisioned that allows scalability via manipulation at the level 5 layer (SIP layer). This allows calls to traverse and pass through the enterprise firewall seamlessly, and with no security issues (as is the case with STUN or manual pinhole techniques).

V1 VoIP SIP Trunks include:

* Managed QoS solution
* Ability to work with any PBX and easy integration with gateways or IP-PBX equipment
* Eliminate the need for expensive and un-flexible traditional voice circuits and reduce the need to purchase additional hardware to interface IP voice to the PSTN
* Cheaper long distance on a SIP trunk when compared to traditional rates
* Complete E911 capability
* Disaster recovery options
* CALEA compliant
* Web based telephone number management
* Detailed call accounting – call history and logs

If you have more questions about SIP Trunks, contact a member of the V1 VoIP team for answers about SIP trunking and what it can do for you.

Article publié pour la première fois le 01/07/2015

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