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5 keys to successful SIP implementation

Christopher Smith, VP of enterprise communications and unified communications, Level 3 | Feb. 18, 2015
Voice over IP uses the session initiation protocol (SIP) to convert phone conversations to data and send it through a public or private IP network instead of using telephone lines or fixed bandwidth T1 and T3 options.

Voice over IP uses the session initiation protocol (SIP) to convert phone conversations to data and send it through a public or private IP network instead of using telephone lines or fixed bandwidth T1 and T3 options. This can be a brilliant way to cut costs, gain flexibility and more efficiently use existing resources, but consider these issues to ensure successful implementation:

* Native SIP. Ask your carrier if their network was designed to deliver SIP end-to-end and the size of their local telephone number footprint. SIP is an open standard protocol used to enable VoIP. Make sure your carrier isn't patching together multiple networks, which may or may not use SIP and could cause quality degradation and make troubleshooting issues more difficult.

A native network is designed to carry SIP because it is made for IP traffic. Non-native solutions use older TDM-based networks, but connect them together with new equipment. Each time data passes through one of these older connection points, quality is impaired and causes jitter or dropped calls. Choosing a non-native SIP network could lead to extra work for the implementation team, a more complex solution for the engineering team to support, and potential installation scheduling delays.

You may also need to invest in additional equipment for a non-native SIP solution. Many enterprises operate with a mixture of older, non-IP premise PBXs and key systems, while other sites have upgraded to IP-capable gear. Having to lease or purchase gateway devices to allow the aging gear to speak SIP will add cost and support complexity, the opposite of what IT leaders are solving for. Some service providers have designed intelligent support at the edge of the network that eliminates the need for extra converter devices. Seek them out.

* Capacity is King. Taking the time to consider all future bandwidth needs will assure quality voice service after implementation. Evaluate and account for all network traffic, including voice data, video and other elements that will now travel over the corporate MPLS network.

Bandwidth is defined as the ability to transfer data (such as a VoIP telephone call) from one point to another in a fixed amount of time. It is measured in terms of speed, latency, delay and jitter. The good news is that in the SIP world, bandwidth can be purchased in much smaller increments than fixed ISDN, in the form of Current Call Paths (CCPs). Some SIP trunking solutions that replace outdated ISDN and TDM access circuits allow call paths to be shared across all enterprise office sites, eliminating waste and cost.

* Enhanced Features. Every good plan starts by thinking about the "what ifs." When you decide to implement SIP-based voice, keep in mind business continuity and disaster recovery (BCDR) plans. SIP based voice services are more flexible and programmable, so make sure your provider has designed inherent BCDR options into their SIP services. These services may not be an up-front offering by your carrier, but they remain a critical element of any voice plan. Are they included in the service, or are there extra costs? Consider:

 

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