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SIP trunking primer

Session Initiation Protocol (SIP) trunking is the use of VoIP to help connect a traditional PBX to the Internet. SIP trunking can reduce communication costs and combine voice, video and data in a single line, allowing for unified communications capabilities. Learn about using SIP trunking services in this primer.

Session Initiation Protocol, or SIP, is a signaling protocol for initiating multimedia usage in a network, including video, voice or chat. SIP establishes the end system to be used and handles call transfer and termination. An important aspect of SIP is SIP trunking, a use of VoIP that can minimize communication costs and maximize network usage with unified communications.

What is SIP trunking and why is it important?
SIP trunking is a service provided by an Internet telephony service provider (ITSP) that uses SIP to connect a traditional private branch exchange (PBX) to the Internet. In traditional telephony, the phone company delivers services over a "trunk" that connects the PBX to the public-switched telephone network (PSTN), and this trunk carries the phone calls from the corporation to the public. With SIP trunking, the physical wires in the trunk are replaced with a converged voice and data network. This eliminates the need for an expensive media gateway and reduces long-distance charges. SIP trunking combines voice, video and data in a single line, eliminating the need for a separate physical medium for each one.

SIP trunking consists of three main elements: the ITSP or SIP trunking provider, the IP PBX, and a border element, which facilitates connectivity among an enterprise IP network, the PSTN, and an external IP-carrier network. The border element, which could be a SIP-capable firewall or a switch to transfer calls in and out of the PSTN, is generally managed by the service provider.

For example, a Los Angeles-based customer service rep at a company employing SIP trunking places a phone call to a client in Chicago. The call is then converted to an IP call (or originates as one) before it leaves the office, travels mostly over the service provider's IP network, then drops onto the PSTN at the termination point. Because most of the call traveled over the service provider's IP network rather than the PSTN, traditional long-distance calling charges won't apply, and the cost of the call will be a fraction of what it would have been if placed over a traditional PBX.

The benefits of SIP trunking
Thanks to this level of cost savings, SIP trunking benefits any enterprise with a PBX that connects all internal users and where the employees make long-distance calls on a regular basis. With the commonly used ISDN, each circuit has a maximum of 30 channels. But with SIP trunking, there is flexibility in line usage because companies don't have to buy capacities -- they can have as many or as few users as they want.

Companies can also save on the costs of media gateways because gateways needed to connect to the PSTN will reside at the ITSP. SIP trunking is also a more efficient alternative to an ISDN or other traditional TDM lines, such as BRI. An ISDN sends digital voice and data over the traditional copper wire, requiring the monthly line rental of ISDN circuits. Because SIP trunks eradicate the need for costly PSTN gateways and traditional TDM lines by using an IP connection to the ITSP, enterprises do not incur costs besides what they pay for the ITSP.

Another benefit of SIP trunking is in bandwidth utilization. Both telephony and Internet lines often utilize bandwidth at a low rate. With the telephony line, telephone usage is generally characterized by several hours a day with many calls and several hours a day with very few calls. Internet usage is rather erratic, with bursts of use throughout the day, thus wasting the capacity at the time that usage is low, for both types of lines. With SIP trunking, combining these to a converged line, you can optimize bandwidth for average consumption rather than peak usage. Quality of service (QoS) settings can allow for prioritizing between voice and data and can ensure that the capacity you need is always available when you need it.

SIP trunking also delivers real-time communications applications – such as IM, presence, video conferencing, and application sharing -- in a cost-efficient, reliable way. With SIP trunking, these real-time communications applications improve collaboration and productivity by reducing human latency and increasing awareness of employee locations, especially for initiating ad hoc meetings. For example, presence allows a hospital administrator to know where the hospital's doctors are at all times, in case of an emergency.

Least cost routing (LCR)
Creative use of SIP trunking can allow an enterprise to reduce calling costs even further, if it employs the use of least cost routing (LCR) and multiple ITSPs. LCR is the process by which a company chooses the path of an outbound communication by the price of the call, opting for the lowest price. If a company uses multiple SIP trunks from different service providers based on geographic locations and time zones, each call can be routed to the cheapest service provider based on country codes, saving a significant amount of money on international calls.

In terms of ROI, this depends on the company's size. Most medium to large companies have multiple locations and often use international calling, so businesses in this scenario can save 50% to 60% of their usual communication costs, and ROI can be achieved within 12 months.

Addressing the challenges of SIP trunking with best practices
With all IP-to-IP connections, security can be an issue because they are always connected to the Internet. SIP trunks are vulnerable to standard signaling and media security issues, as well as peer-to-peer issues if enterprises trust others to provide authentication. However, SIP server and proxy technologies can effectively manage the flow of SIP traffic, as long as there is a reliable IT manager to deal with the security threats with verification and authentication policies.

Another hurdle with SIP trunking is deploying and managing equipment from different vendors. As in any unified environment, mixing equipment based on the SIP protocol from different vendors can cause interoperability problems stemming from incompatibility. Two IP PBXs may seem to be compliant to the same requests for comment (RFC), but because of the open-editing nature of RFCs, the RFC definition for SIP has become overly flexible and vague.

To help battle this problem, SIP Forum, an IP communications industry association, has introduced SIPconnect. This is a recommended set of industry interoperability guidelines designed to facilitate the connection between SIP-enabled IP PBXs and SIP-enabled service provider networks and to establish industry-wide norms. It specifies VoIP protocols/features, a reference architecture and implementation rules. By using SIPconnect, enterprises can solve interoperability problems.

For more information on SIP trunking:
SIP trunking ROI: Linking VoIP islands and more
The ROI for SIP trunking goes beyond linking VoIP islands. It also presents additional cost-saving scenarios by bringing enterprises beyond the voice-centric IP PBX and reducing/eliminating the dependency on the PSTN. In this tip, learn about SIP trunking ROI, including bridging VoIP islands, telephone services, hardware, centralization and convergence as they relate to different business scenarios.

 SIP trunks a no-brainer for VoIP rollouts
Learn what a Session Initiation Protocol (SIP) trunk is and how it can strengthen the value of your VoIP and unified communications deployments.

 What are the benefits of SIP trunking over T1 trunking?
In this expert response, learn why SIP trunking is preferable to other implementations.


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