Real-time IP facsimile: Protocol and gateway requirements

In an attempt to integrate telephony and data networks into a single analogous framework, techniques have been developed to enable telephony signals to be packetized and transmitted over Internet protocol (IP) networks. Although the largest potential contributor to telephony traffic over IP is voice data, facsimile is a widely used service and must therefore be considered as an important feature of the IP telephony framework. It is argued that facsimile and voice traffic not only have different properties, but also different requirements for service implementation. Consequently, it is proposed that in order to provide an IP telephony system (including both voice and facsimile) with a given quality of service, the mechanisms behind transfer of facsimile information shall differ from those for voice. Many of the requirements of an IP telephony system can be found in ITU-T Recommendation H.323. Covering a broad spectrum of possible media applications, the recommendation's generality is complemented by its recognition that different media streams require different transport mechanisms. For instance, H.323 defines separate transport protocols for data and voice media streams: voice streams are to be transported using real-time protocol (RTP) while data streams are to be transported using transmission control protocol (TCP). This paper presents a descriptive analysis of the differing requirements for reliable voice and facsimile media transfer over IP networks. These requirements lead to a set of protocol and gateway characteristics that enable high-quality facsimile transfer over packet-oriented networks such as the Internet.