10,000-Seat wVoIP Deployment in Japan

By Ted Stevenson | Jul 25, 2005 | Print this Page
http://www.enterprisenetworkingplanet.com/unified_communications/10000-Seat-wVoIP-Deployment-in-Japan-3522751.htm

Meru Networks—U.S.-based vendor of centrally managed wireless LAN infrastructure—was founded with the mission of bringing robust VoIP capabilities to wireless networks. Over the next 18 months, the company will put its products to work for Osaka Gas, one of Japan's largest public utilities.

Once complete, the network will connect some 10,000 Osaka Gas staff, using voice over 802.11b wireless LAN indoors, and handing off to NTT DoCoMo's cellular service outside the office. The system will employ NTT DoCoMo's dual-mode Wi-Fi/cellular handsets.

"Handoff" in this context is not the "seamless" handoff, long envisioned by futurists, and now undergoing trials in Europe and the United States. "It's more of a hard handoff," Meru director of product marketing Joel Vincent told EnterpriseVoIPplanet. "You may need to redial your call," Vincent said.

Still, the design achieves the vision, much discussed within the industry, of using the comparatively inexpensive IP network where it is available (and mobile wireless service quality is often poor) and relying on the cellular network when out of reach of the IP net. According to Vincent, the seamless handoff capability will be added later.

Osaka Gas selected Meru after extensive competitive trials involving other high-profile U.S. infrastructure vendors. The company credits its new Voice Services Module (VSM) software (also officially announced today) with the success of its bid for the project. VSM is designed to ensure high call quality in three ways.

The module does this first, by placing management controls on the number of calls being handled by any network access point or "virtual cell" (a construct of Meru's proprietary architecture). This "Call Admission Control" is a fairly straightforward way to manage quality—the more voice calls and data streams running over a given access point, the less reliable the call quality. When the limit is reached, new incoming and outgoing callers receive a busy signal or message explaining the situation.

Second, working in concert with Call Admission Control is a dynamic Load Balancing capacity. Calls can be switched among access points in real time—and among virtual cells during the call setup phase -- as calls terminate and connection patterns shift. In this way phone traffic can be optimized across the entire network fabric.

The third component of the Voice Services Module is Dynamic Error Correction. The most likely manifestation of quality issues in VoIP is dropped packets, which result in tiny "holes" in the sound stream. VSM combats this effect by replicating the previous packet whenever one is dropped, and resending that. While this doesn't eliminate the interruption of the sound stream, the result is far less disconcerting, according to studies.