Abstract: Methods for providing improvement in Voice-over-IP communication systems, and hardware for implementing the methods, are disclosed. A first aspect provides a method of improving on the efficiency of RTP used to transport VoIP voice calls by reducing the overhead of second and subsequent calls on a link to almost zero using trunking. A second aspect uses bandwidth awareness to compress RTP payload data captured from the network. This involves capturing G.711 encoded RTP data directly from the network ( as opposed to at source ) and transcoding that data in such a way as to take account of the available bandwidth on an outbound link. A third aspect uses dynamic and transparent packet fragmentation and reassembly based on RTP interval to reduce VoIP latency and jitter. A fourth aspect uses dynamic re-writing of SIP messages to provides automatic fail-over and load balancing of SIP servers. This involves capturing SIP call set-up messages and re-writing and duplicating them to direct them to multiple servers.