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We propose a novel Voice over IP (VoIP) traffic detection approach which can penetrate some important barriers in front of common traffic identification mechanisms: encryption, complexity and camouflage. First we use the information source, channel and destination model in the communication theory to reexamine our understanding of Internet data flows. The packet, the Inter Arrival Time (IAT) between 2 successive packets and the payload length which are only of statistical meanings previously, are now assigned physical properties as pulse, pulse arrival time and the amplitude. The packet-based Internet traffic is therefore transformed to a discrete-time pulse sequence.We then use spectral analysis to exhibit frequency domain behaviors of the general Internet background traffic and some typical VoIP traffics. The result reveals that all kinds of VoIP traffics have their own characteristics (i.e., peaks in certain frequencies) on the power spectrum. Experimental data on simulation as well as real life data set indicates that the approach is effective on recognizing VoIP traffics. We argue that the novel approach complements existing traffic identification mechanisms. The rationale behind it can also be applied to other fields.

We introduce the current status on the development and trails of ATM network management in China putting emphasis on TMN interfaces. The following items are addressed: standardization of ATM network management, experiment system development, and trails with ATM network elements.

A new ATM signaling system architecture, which is called SCCA, is proposed and a possible configuration of the signaling system for a campus ATM network is given based on SCCA. This new signaling model has a centralized CAC & routing server to provide some central work of signaling and lets the separated signaling stack software, which is called the SIGStacker, maintain the ports' signaling stacks. Because all the SIGStackers are running on externally attached workstations and these SIGStackers are configurable, the signaling system can be feasibly configured. Therefore, any number of SIGStackers and the same number of workstations on which SIGStackers run can be configured for a switch. A SIGStacker can manage signaling stack of any ports on a switch. Users can easily find a compromise between cost and efficiency of the signaling system. SCCA is also convenient for management. In SCCA, the CAC and routing server share a common database with the CMIP management agent, so the controlling plane of the switch can be conveniently managed by the management plane and furthermore by the central management entity.

Efforts are underway to enhance the Internet with quality of service (QoS) capabilities for transporting data. The reservation protocol (RSVP) provides a signaling mechanism for end-to-end QoS negotiation. Hence, the performance of TCP, a broadly used transport protocol, over RSVP becomes a topic of intense interest. A modified RSVP called bi-directional RSVP (bi-RSVP) is proposed based on the analysis of TCP over RSVP. The bi-RSVP provides resource reservation for ACK packets to improve the TCP performance. The simulation results testify to the effect of the implementation of bi-RSVP for the TCP stream in some network environments.

The research on end-to-end congestion control scheme has been developed widely. But the end-to-end adaptation scheme naturally has some limitations, such as the probing problem and fairness problem. In order to solve these problems, the assistance of network is necessary. Therefore, a network-assisted congestion control (NACC) algorithm is presented. The network-assisted congestion control algorithm is quite simple, but very efficient. Additionally, the NACC algorithm could provide a TCP-friendliness. Finally, the results of simulation prove that the NACC algorithm could fulfil the max-min fairness and TCP-friendliness.

We present a variable additive increase adaptive algorithm, which utilizes RTP/RTCP as the protocol to collect QoS information. Compared with the constant additive algorithm, the flow enhanced by the variable increase algorithm is more stable. We also simulate the algorithm and analyze the results.

We present a TCP-friendly flow control algorithm that utilizes RTP/RTCP as the protocol collecting QoS information. The current multimedia source has the requirement of granularity with which the sender can change its transmission rate. The algorithm presented allows for adapting the transmission rate of the multimedia source in a TCP-friendly way and also takes the restrictions imposed by the multimedia source into account at the same time. Finally, we simulate the algorithm and analyze the results.