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eWeek Labs recently got an earful of voice over IP. Working with VOIP equipment from industry leaders Avaya Inc. and Nortel Networks Ltd., we discerned the strategic decisions IT managers can make now that will enable them to successfully bring VOIP to the enterprise and improve data network performance at the same time.
Putting voice traffic on a data network can significantly lower capital costs by unifying cable plant and network equipment. It can also lower operational costs with distributed call center agents and the enabling of self-administered conference calling and toll-bypass outside the United States.
IT managers who converge data and telecommunications traffic—and likewise merge the IT and telecom departments—will likely reap the rewards of significantly lower add/move costs associated with telephone handsets, staff efficiency gains, easier deployment of new calling features and increased network reliability for data applications.
However, VOIP is also fraught with barriers, and one of the biggest is the legendary reliability and quality of current phone systems. Add to that rock-bottom tariffs for domestic calling, the lack of a VOIP-only “killer app” and—most significant of all—the cost to upgrade data networks to carry voice traffic, and most organizations can find little reason to immediately switch.
However, whats new now is the availability of advanced VOIP applications, which will be the telephony growth platform of the future. This means everyone from station users (telecom parlance for a user of a telephone handset), to call center agents, to executives and their assistants to reception operators and telecom managers can use VOIP equipment today with the same features they get from the current phone system.
During testing of Avayas and Nortels telephony equipment, Category 5 data cabling provided dial tone and power for handsets. Call quality using both vendors systems was, for the most part, indistinguishable from standard digital telephones.
To get to this point, however, most data networks will have to undergo substantial remediation—both in terms of adding equipment to supply power over Ethernet and implementing protocols in switches and routers. For example, both Avaya and Nortel use the IEEE 802.1p protocol (which defines how multicasts should be filtered at Layer 2 to ensure that they are not propagated over Layer 2-switched networks) and 802.1Q (for virtual LAN tagging).
To evaluate not only VOIP products and infrastructure but also the process organizations must go through to determine the technologys return on investment, eWeek Labs developed an RFP (request for proposal) for a fictitious company called Industry Inc.
We worked with Avaya and Nortel to fine-tune the RFP, which detailed a two-phase voice implementation (see chart).
The first phase, a large pilot project, required that the vendor support 1,500 VOIP handsets with equipment available today. The second phase specified supplying 10,000 VOIP handsets—enough to outfit the entire Industry user base—with technology available today and/or slated for release within the next year. The ballpark cost estimates, assuming numerous variables, are $2 million for Phase 1 and $7.9 million for Phase 2.
Our requirements were guided by the principle that the VOIP implementation must provide service equal to or greater than the phone system its replacing. For example, handsets must deliver dial tone even in a power outage, and audio quality must equal that of a traditional business telephone. Furthermore, management features used by telecom staff to maintain the system must meet or exceed those already available. (For more VOIP requirements, see chart.)
In terms of network security, one big advantage that traditional telephone systems have over VOIP is that each station is wired directly back to the PBX. With that said, the phone systems we evaluated took security very seriously and are on firm footing to secure the system from attacks and abuse. However, as VOIP systems become more widely deployed, weaknesses will be found. IT managers will have to be particularly vigilant about the performance and security of voice applications in the enterprise.
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Avaya
Avaya responded to the RFP with a set of products and services from the companys ECLIPS (Enterprise Class IP Solutions) portfolio. Avayas RFP response (and our subsequent work with its telephony equipment) confirmed that the ECLIPS system will work with nearly any vendors data networking equipment. During an on-site visit to its research and development facility, in Denver, we observed Avaya telephony equipment being tested with a wide variety of data products, including products from industry networking heavyweight Cisco Systems Inc.
IT managers should ask about interoperability among any potential VOIP providers systems and the data equipment already installed in the enterprise. In most cases, there are few technical reasons for sticking with the same vendor for voice and data, although IT managers can be sure they will hear otherwise from sales representatives.
Like most of the major VOIP vendors, Avaya offers a broad range of services—from design and planning to implementation, management, maintenance and support.
Expect—no, demand—extensive consultation with whichever vendor is selected. Savvy chief technology officers and IT and telecom managers will balance the vendors recommendations with the real-life experience of telecom staffers and station users needs. We found that vendors often got all “blue sky” on us, with promises of vastly improved telephony applications. We kept coming back to our RFP to stay grounded in the business objectives of our VOIP evaluation.
Avayas MultiVantage call processing software is used on the companys S8700 Media Server to centralize voice and data operations while also distributing voice application features across the network to all locations, including the call center, manufacturing facilities, branch and home offices, and mobile users logging in from hotels and other remote locations. MultiVantage provides user and system management functionality, call routing, and application integration.
Our work with the Avaya ECLIPS system showed that focusing on the MultiVantage software was critical to understanding the upper limits of the Avaya system.
And those limits are impressive. A single Avaya S8700 that was listed for $20,000 in the RFP response, along with the $25,000 MultiVantage software, can support 300,000 BHCCs (busy hour call completions) in a general, mixed-call environment. The Avaya system also can support as many as 12,000 IP stations, well exceeding our specifications.
Of course, the S8700 Media Server must be equipped with additional Control LAN cards, Media Processor boards and additional software licenses to achieve this capacity.
Avayas Media Servers, including the S8700 used for this project, are Linux-based and optimized for the MultiVantage call processing software and for security. None of the recently announced Linux-based worms, including Slapper, would have been successful against an Avaya Media Server.
Media Gateways, which are integrated into the system, provided connectivity to the public switched telephone network and switching from packet to TDM (Time Division Multiplexing) end points. TDM is the transmission and coding standard used by all current circuit-switched PBX systems.
Avaya uses QSig, or Q Reference Point Signaling, a long-standing telephony standard from the ISDN Private Network Systems Forum designed to allow different PBX systems to interoperate. QSig thus provides seamless communications among the various call centers and headquarters detailed in our RFP.
In our evaluation, the important standards we wanted to see supported included H.323, which covers voice and video transmission over packet-switched networks. We also looked for support of H.323s G.711 and G.749, which govern voice codecs. In addition, Power over Ethernet 802.3af should be on any check-off list (see Tech Analysis, at www.eweek.com/links), as should QSig, RTP (Real-Time Transport Protocol), SNMP, DiffServ and RSVP (Resource Reservation Setup Protocol) to handle call setup over the WAN.
IT managers should also press potential vendors to lay out their plans to support and implement Session Initiation Protocol, an Internet Engineering Task Force signaling protocol for establishing real-time voice and conference calls over IP networks. The standard is still in development, but its interesting because of its potential to exceed the established, but more limited, H.323 call routing and signaling protocol.
Although we did not request it, Avaya pointed out that we could have equipped each of the remote Media Gateways (in this case G700s) with an S8300 Media Server running MultiVantage software in Local Survivable Processor mode. This would have provided added reliability in the event of a WAN failure. We recommend that IT managers add this capability to their RFPs for any business-critical site.
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Nortel
Nortel built its response to our RFP around the Succession CSE (Communications Server for Enterprise) 1000 IP PBX platform. Nortel brought a fully functioning demonstration unit to eWeek Labs San Francisco offices, and we used the system for two weeks to gain hands-on experience.
The Nortel Succession CSE 1000 consists of three building blocks: core hardware, including the Nortel Call Server, Signaling Server and Media Gateways; IP Terminals and client devices, such as handsets and operator consoles; and applications. We think the Nortel system provides a good evaluation model for any VOIP product: Look at the hardware, software and applications to ensure that the product platform fits the bill.
The Succession CSE 1000 Call Server and Signaling Server use the IP network to manage connections between end points, such as LAN- and WAN-based Media Gateways and IP clients. This means the Succession CSE 1000 provides call control services such as address translation, call admission control, bandwidth control, call authorization and H.323 zone management.
The Succession Media Cards inside the Media Gateways support a variety of codecs, including G.711 and G.729A/ B, that bridge packet-based IP and TDM telephone networks. Media Gateways also can be distributed across the WAN, and in this configuration each can support as many as 400 “survivable” IP phones at the remote location.
Both Nortel and Avaya provided a wide range of handsets to satisfy our RFP. The good news is both companies provide several handsets that use power over Ethernet and user stations still look and act like traditional telephones. The bad news is that several handsets use a separate AC power supply, which means they stop working when building power is lost.
Even handsets that use power over Ethernet are hampered by the power restrictions imposed by the standard: Displays were dim and easily obscured by glare—and thats just the LCD monochrome displays.
Nortels Symposium Call Center IP product allows call center agents to be sitting anywhere a company has an office, not necessarily in a dedicated call center. This is an option that is difficult to achieve using a traditional telephone system but quite doable with VOIP.
The Nortel IP Phones we used in our tests provided good voice quality, with none of the echo or “underwater” sounds weve heard in previous VOIP implementations. The handsets rely on Nortel Succession system software, which provides more than 450 features.
Nortels development direction at the desktop goes far beyond the requirements of eWeeks RFP and seeks to incorporate video and file sharing along with user preferences and presence, which affects call coverage rules.
Nortels system also includes a failover option, which performed well in tests.
Senior Analyst Cameron Sturdevant can be contacted at cameron_sturdevant@ziffdavis.com.
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The Scenario
“Industry Inc.” has the following requirements for a new business telephone system that it intends to roll out in two phases.
- Phase 1 1,500 Ethernet-connected handsets
- Phase 2 (one year later) 10,000 Ethernet-connected handsets
RFP Details
1. Are the handsets line-powered?
2. What power standards (or developing standards) or protocols are adhered to? For example, 802.3af?
3. Describe the call processor functionality.
- How many systems, using which cards, are necessary to support 1,500 IP handsets?
- Describe how the system can be scaled to 10,000 handsets.
- Describe how phone management is distributed across servers. Specifically:
•Under what circumstances do you recommend concentrating all
voice mail at a central location?
•Under what circumstances do you recommend distributing voice
mail to branch offices/remote factories?
•Given that Industry wishes to consolidate as much telephone-
specific administration as possible to headquarters, describe
what services must be administered locally. - Do you provide automatic route selection/least-cost routing?
- Detail the pre- and post-sales traffic analysis tools, including traffic reports and services, that assist IT staff in gauging:
•Failover routing plans
•Determining factors such as BHCA (busy hour call attempts), BHCC
(busy hour call completions), CPS (calls per second)
•Capacity planning for future growth - What parameters can be taken into account when sampling traffic load to determine offered and carried load?
- How many classes of service do you provide for toll restriction?
- Can your equipment provide classes of service based on a PIN?
- Can you restrict individual numbers from being called?
- If call quality on the IP network degrades, what options are available to put calls through on the traditional telephone network?
4. Detail how the new system will access the public switched telephone network.
- Describe the equipment necessary to provide digital access for 150 outgoing lines to the PSTN when 1,500 IP handsets are in use.
- Describe the equipment necessary to provide digital access for 750 lines to the PSTN when 10,000 IP handsets are in use.
- Describe the equipment necessary to provide digital access for 150 incoming lines for DID (Direct Inward Dial) service in the event that 1,500 IP handsets are in use.
- Describe the equipment necessary to provide digital access for 750 incoming lines for DID service in the event that 10,000 IP handsets are in use.
- Explain how the call processor is integrated with the PBX.
- Can multiple PBX servers equally share multiple call processors?
- Can individual trunk groups transparently span multiple call processors?
5. Describe how your proposal addresses integrated messaging.
- Use a diagram to demonstrate the connection of your voice mail system to an e-mail system. Explain what is necessary for the e-mail client and server software to retrieve voice mail messages. Also explain, if applicable, how your voice mail system can retrieve e-mail text.
- List the e-mail systems that integrate with your product.
- How does your voice mail system generate subject lines for the e-mail system?
6. How do telecommuters fit into the VOIP system?
- Describe features that enable employees to have an “in the office” telephone experience from home, hotel room or other remote location with Internet access.
- How are these services secured?
7. What are the target numbers for acceptable network bandwidth to support VOIP? What additional protocols should be implemented to support VOIP?
- Provide the amount of peak bandwidth necessary per phone per call between the handset and call processor.
- List and detail any networking services that must be implemented, such as Dynamic Host Configuration Protocol, Domain Name System, etc.
- List and detail the QOS (quality-of-service) standards supported by the handsets, call processor, gateways, network switches, other network equipment and end nodes, where applicable, that ensure end-to-end voice service priority. Explain how the components implement QOS standards such as 802.1p, DiffServ, TOS (Type of Service) and RSVP (Resource Reservation Setup Protocol).
8. Call accounting features: Describe, or show an example of, the following features:
- SMDR (Station Message Detail Recording)
- Busy reports on outbound trunk groups
- Busy reports on incoming trunk groups
- Usage reports based on PIN or account numbers
- Other reporting capabilities
9. List all significant standards that are currently supported by your VOIP telephone system. Describe the ability of your VOIP telephone system to support SIP (Session Initiation Protocol), MGCP (Media Gateway Control Protocol) and other significant protocols and standards.