One of the major obstacles facing designers of video conferencing systems is the problem of ameliorating the effects of congestion on interconnected packet-switched networks that do not support real-time communication. We present a framework for transmission control that describes the current network environ- ment as a set of sustainable bit and packet transmission-rate combinations and show that adaptively scaling both the bit and packet-rate of the audio and video streams can reduce the impact of congestion. We empirically demonstrate the validity of adapting both packet and bit-rate using a simple feedback mechanism and simple adaptation heuristics to deliver audio and video streams suitable for low-latency, high-fidelity playout.