Skip to Main Content
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) operating at 11 Mbps for example, can support only about 12 G.711 voice connections with a 20 ms packetization interval. These effects can be mitigated by taking into account the available latency margin of the call and using it in the VoIP parameter selection. In this paper we propose the use of an adaptive voice packetization server (AVP-RTS) which splits the RTP VoIP connection into two legs. In this way each end of the call is negotiated separately and the server can allocate the available latency margin (and the ensuing capacity gain) asymmetrically across the call. We propose new algorithms for performing this capacity assignment and compare them to the conventional voice packetization scheme. Results from extensive simulations show that by using the AVP-RTS server we can significantly improve the multi-AP VoIP capacity for certain typical IEEE 802.11 situations.