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F E A T U R E

2003 Survivor's Guide to Digital Convergence

December 6, 2002
By Sean Doherty

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	In this article
	Introduction
	Bandwidth Requirements
	Content Delivery
	Companies to Watch
	Standards
	Enterprise Speed Limits
	Web Links

The question to ask about digital convergence is not whether the merging of voice and video traffic networks will occur, but when it will occur. Even if you think the answer for your organization isn't quite "now," it's bound to be "very soon." Converting separate circuit-switched voice and video traffic "overlay" networks to packet-based technology makes sense. You'll save money by standardizing your applications on IP-enabled networking gear and having a common platform for all network management. But there's more to involved than using audio- and videoconferencing to reduce travel costs. This shift can become a strategic revenue-generating initiative when you combine call-center technology with CRM (customer relationship management) over IP and enable unified-messaging solutions for mobile employees.

Although the telecommunications industry took a nosedive in 2001, shipments of convergence components such as IP PBXs increased as many enterprises moved toward convergence (see "U.S. PBX vs. IP PBX Line Shipments").

CTI (computer-telephony integration) equipment also came into more widespread use as enterprises invested in call-center technology to attract and retain customers in 2002; that category is projected to rise by 13 percent per year (CAGR) from 2001 to 2005. And as more employees work remotely over wireless and broadband technologies, such as cable and DSL, digital convergence will bring them all the benefits of the office while they're on the road or at home (see "U.S. Broadband Services").

Unified messaging solutions also continue to grow, with a projected 32 percent CAGR from 2001 to 2005--further evidence of this category's viability.

Not all enterprises have found the reasons for convergence compelling enough to warrant purchasing new equipment to replace older, but working, equipment in the near term. Enterprises with PBXs and videoconferencing systems in production may not find a sufficient incentive to migrate these services to their IP networks in 2003. Your systems may still have features you've never seen. And they may still be depreciating in value. Replacing them and writing them off would not sit well on your bottom line. But if you need to replace legacy equipment, if merger takes your enterprise beyond the maximum limits of support, or if you're relocating to a new building, you must invest in technology with IP support.

Whether your enterprise is on the road toward convergence or still in the planning stage, you need some assurances that converged voice and video applications over IP will benefit customers and employees and not burden them with long delays and intermittent network outages. When convergence happens, your IP network must be ready.

Convergence: Ready or Not

IP-enabled equipment and applications have become a dominant force on the Internet and in Web services. IP provides an adequate transport over Layer 2 network protocols, such as ATM, Ethernet, frame relay and ISDN, as well as in Layer 1 networks using ADSL and cable. IP's supremacy is rooted in the simplicity of its best-effort attempt to deliver a packet to its destination. A best-effort protocol, however, may not be adequate to deliver real-time voice and multimedia packets.

Before you consider convergence, you need to evaluate your network's service levels. We recommend using four indicators: bandwidth, delay, jitter and packet loss. If your network is barely meeting its service levels, you'll need to provision more bandwidth and consider distributing the available bandwidth using subnets, VLANs (virtual LANs), traffic-prioritization equipment and content-delivery technologies to help you meet service levels and keep voice and data traffic flowing. In general, bandwidth should be as large as possible while minimizing delay, jitter and packet loss.

Bandwidth Needed for Common Coding Algorithims

click to enlarge

When you apply this rule in the real world, remember audio and video traffic is primarily transmitted over UDP, which, unlike TCP, does not retransmit lost packets. In the new network, the show must go on, and there's no place for any dropped or out-of-sequence packets. Long network delays result in stopgap conversations with extensive silent periods between sentences. Network jitter, the variation in end-to-end delay and sequential packet delivery, results in jerky video and stuttering audio. The same result occurs when intermediate devices drop or lose packets because of congestion.

The ITU (International Telecommunication Union) standard G.114 recommends that network transmission time should not exceed 150 ms (milliseconds), including the delay in equipment processing time and the propagation delay in traversing the network. In the real world, satisfactory performance for audio and video continues with up to a 300-ms delay, but as 400 ms approaches, quality deteriorates. If network delay is a problem, identify the congestion points and segment them into subnets or VLANs to share the bandwidth. Or provision more bandwidth and add the capacity necessary for convergence.