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Migrating to VOIP

Planning for an IP-oriented infrastructure is crucial when moving over to VOIP.

Dave Paulding
By Dave Paulding, regional sales director, UK, Middle East and Africa, for Interactive Intelligence.
Johannesburg, 30 Jul 2010

One of the most crucial steps any organisation should take when deploying a solution for voice over IP (VOIP) is to plan for an Internet Protocol-oriented infrastructure.

True, VOIP deployments vary significantly from one organisation to the next, and migrations don't all follow the same path. But by understanding the principal elements of VOIP - network bandwidth and CODEC requirements, quality of service (QOS) for networked voice traffic and standards such as the Session Initiation Protocol (SIP) - organisations can establish a specific and effective baseline for their VOIP solution deployment and ensure the best possible system and network performance.

Critical phases of VOIP planning and infrastructure design:
Since the mid 1990s when voice over IP was introduced, the IP industry has increasingly turned to open standards like SIP and recommended CODECs for network bandwidth in the effort to improve VOIP network readiness and security worldwide. At the same time, to make migrating to VOIP more straightforward, IP vendors and service providers have continued to establish essential planning and design functions for a successful migration.

Six key functions

Plan the right architecture for the company's particular VOIP deployment model:
Is there a single site or distributed locations? Is the migration phased, ie, moving only selected systems, departments or sites to VOIP, or across the entire enterprise? Whichever model and migration approach, the planning goal is to structure the organisation's cable plant design and data centre resources sufficiently for VOIP call processing to all potential users.

Understand the factors that impact voice (call) quality:
As voice transmission travels over an IP-based data network, the clarity and quality of the call can be negatively impacted by delay, echo, and jitter. Delay stems from the amount of time it takes a VOIP voice packet to be created, sent across the network and converted back into sound, while echo results from delay(s) in any point of the voice packet process. Jitter occurs when voice packets arrive at an interval greater than they're sent. Overall, echo becomes more noticeable as delay increases, and jitter is more prevalent when an IP network provides different waiting times for voice packet transmissions, or varying levels of latency.

When planning the network for VOIP, note that delay has the most impact on voice quality since it precedes echo and jitter. To achieve the best potential quality for a VOIP call, a general guideline is that one-way delay should not exceed 150 milliseconds. A range of 150-400 milliseconds is acceptable for higher one-way delay ranges, provided system administrators are aware of the increased transmission time and its impact.

Actual network prep is usually better left to a vendor or consultant certified in network assessments.

Dave Paulding is Interactive Intelligence's regional sales manager for UK and Africa.

Analyse and prepare the network for voice and data:
Analysing a network's voice and data traffic volumes and planning the appropriate capacity for VOIP isn't something a company's IT team does on a routine basis. Therefore, actual network prep is usually better left to a vendor or consultant certified in network assessments.

Determine CODEC and bandwidth needs:
Defined, CODEC is the COmpression/DECompression that voice-based data packets experience when they're converted from analogue form to digital signals for VOIP. CODEC factors can originate in a PBX/IP PBX phone system and be shared by analogue phones, or take place in phones themselves.

In general, each VOIP voice packet in a call transmission contains 40 bytes of IP overhead, and overall for CODEC bandwidth with overhead, combined WAN data (voice, video and data) should not exceed 75% of available link bandwidth if planning to optimise the network for VOIP.

Determine QOS priorities and the appropriate methods/policies:
Another key component of VOIP network planning is deciding where the organisation's QOS priorities lie. Using the example of a multi-site centralised call processing deployment model, where call processing originates from a central site and reaches multi-site locations via SIP tie lines to a WAN (or MPLS), QOS can reside in network points for campus access, campus distribution, the WAN, and branch locations.

Noting those points and having determined QOS priorities, the next steps are to characterise the data network, implement QOS policies and monitor the network's operational load. QOS priorities themselves should consider how the network will be used and what level of network service is required (integrated services, differentiated services for guaranteed latency/delivery, best effort).

Address security needs and potential issues:
Any organisation that handles confidential information on an IP-based network must make securing calls and data an ongoing priority. Fortunately, the security mechanisms now available for IP technologies are some of the most stringent ever, and new standards are constantly being deployed to make security even more concrete.

Among these standards, the Session Initiation Protocol (SIP) is highly accepted worldwide for its rigorous message encryption and user authentication in a VOIP environment, in large part because SIP is regulated by the Internet Engineering Task Force (IETF) for IP communications security.

The more a business understands upfront about VOIP and how it works, the more straightforward it's migration to IP communications will be. And by knowing how the details of VOIP can affect system and network performance both positively and negatively, a company will be better prepared to optimise VOIP performance throughout the organisation after deployment.

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