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How To Use VoIP On Your Wireless LAN: Page 2 of 7

Microsecond predictability of synchronous cellular systems is conducive to a synchronous sleep-wake scheduling discipline for the hardware and firmware in the handset. The 802.11 universe, instead, uses CSMA contention methods that conspicuously lack centralized synchronous timing. This is the principal strength of 802.11, and can be viewed as yet another replay of the perpetual debate between packet-and circuit-switching, between Ethernet and ATM, and between robust/adaptive channel access (good enough) and rigid timing (perfection). This means that new protocol engineering is needed to develop power-save timing techniques that work in the non-synchronous 802.11 world.

It's possible to change the 802.11 MAC into a synchronous, slotted-TDMA design either on a full-time or part-time basis. Numerous proposals and proprietary implementations do exactly that. But the result would no longer be Wi-Fi. Although a technically valid approach, a globally synchronous Wi-Fi infrastructure would be incompatible with existing 802.11 devices, and in some cases, not compatible at all. For the immediate future, we must concentrate on how to work with VoIP and Wi-Fi as it's understood today. That means taking full advantage of the toolkit of new features produced by the TGe QoS group and beginning to be certified as WMM (Wi-Fi Multimedia).

VoIP profiles
VoIP is a constant bit rate (CBR) application. VoIP packets, or frames, are continually generated at a constant interval, usually 10, 20, or 30 ms, although there are exceptions (22.5 ms). The CBR frames travel from source to sink, passing through a various equipment and links along the path. ITU-T Recommendation G.114 specifies an end-to-end latency budget of 150 ms or less. If there's a wireless LAN at the source and/or sink, each WLAN can have only a small portion of the 150 ms. If the CBR packets traverse the Internet or a busy corporate network, the arrival timing at the sink won't faithfully replicate the injection timing at the source. Packets will arrive late, or sometimes not at all. And packets may arrive in bunches at well-timed CBR intervals.

Internet-style latency, jitter, packet loss, and bunching are problems for older codecs. The legacy codecs are far less tolerant of packet loss and jitter than modern codecs designed for Internet use. Some would say the classic codecs are largely intolerant of sub-optimal channels. That's understandable given their history. It's s also understandable that there's interest in tweaking wireless LANs to support the stringent needs of legacy codecs. However, that becomes less important with the proliferation of new codec technology.

In the world of VoIP over broadband Internet, conditions are far less than synchronous or optimal. This has spurred development of advanced codec designs that compare favorably with high-end ITU specifications, such as G.729. For example, the iLBC codec from Global IP Sound is now mandatory in the CableLabs PacketCable spec, is an experimental track specification (RFC 3952) within IETF, and is the basis for at least one well-known Internet VoIP product (Skype). This codec claims to withstand 30% packet loss while maintaining voice quality in the presence of Internet-like delay and jitter. It seems to be the perfect answer for a non-synchronous, open system like 802.11.