WLAN VoIP capacity allocation using an adaptive voice packetization server

摘要

WLAN VoIP capacity is known to be very low due to the effects of overheads at various protocol layers. An IEEE 802.11b access point (AP) for example, can support only about 10 G.711 voice connections using a 20 ms packetization interval without advanced header compression. This poor performance can be substantially improved if the connection latency margin and packet loss performance are known in advance and are used in the selection of RTP packetization parameters. In this paper we take this process one step further and investigate the use of an adaptive voice packetization server (AVP-RTS) which splits the RTP VoIP connection into two separate call legs. In this way each call leg can use different packetization parameters, thus assigning the capacity gain asymmetrically across the connection. Algorithms are proposed and compared for performing this assignment. We show that by using the AVP-RTS server we can significantly improve the multi-AP VoIP capacity for certain typical WLAN situations.