Packetized voice transmission using RT-MAC, a wireless real-time medium access control protocol
ACM SIGMOBILE Mobile Computing and Communications Review
Voice transmission in an IEEE 802.11 WLAN based access network
WOWMOM '01 Proceedings of the 4th ACM international workshop on Wireless mobile multimedia
Novel Multiple Access Protocol for Voice over IP in Wireless LAN
ISCC '02 Proceedings of the Seventh International Symposium on Computers and Communications (ISCC'02)
Survey on QoS Management of VoIP
ICCNMC '03 Proceedings of the 2003 International Conference on Computer Networks and Mobile Computing
Convex Optimization
A multiplex-multicast scheme that improves system capacity of voice-over-IP on wireless LAN by 100%
ISCC '04 Proceedings of the Ninth International Symposium on Computers and Communications 2004 Volume 2 (ISCC"04) - Volume 02
A multiplexing scheme for H.323 voice-over-IP applications
IEEE Journal on Selected Areas in Communications
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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 20ms 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.