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In VSC-HVDC converter station, the level of electromagnetic interference at the converter station directly affects the control system and the protection system. Every fast firing and breaking process of the IGBT leads to an electromagnetic transient process, as well as an electromagnetic field with high frequency and wide spectrum. In this paper, conducted electromagnetic interference on both sides of AC / DC converter station was studied; the radiation electric field near the single-stage valve tower copper bar and PCB board are measured and analyzed.

In order to predict accurately the radio frequency interference produced by HVDC voltage source converter(VSC-HVDC) valve, based on a ±320kV HVDC modular multilevel converter (MMC-HVDC) station in China, the calculation model was established using the method of moments. Electromagnetic interference characteristic was analyzed by the model in this paper. The near electric field distribution and far field propagation were calculated in different frequencies. Finally, the comparison between the electromagnetic calculated results and the RFI limits of HV and MV substations at 200 meters distance was provided, which indicated the electromagnetic calculation results satisfied the limits. The results can be adopted as a reference for the electromagnetic compatibility design and electromagnetic interference protection of MMC-HVDC converter valve.

Cloud computing, a new thing which is only three years old, becomes the most fashionable technology and developing trend in IT industry now. People say “X as a service” in the cloud computing, which means that everything including software and hardware can be taken as a service for consumers in the cloud. Meanwhile, a high quality multimedia conferencing has been an urgent requirement these years. The products in the market are immature. They can't support very large number of participants and multiple HD video in a meeting. In this paper, we consider conference as a service, and propose a feasible framework of multimedia conferencing in cloud. As it follows the structure of SOA, we name it Service-Oriented Cloud Conferencing. We then present the software structure and flow procedures. We prove that the cloud conferencing overcomes the deficiencies of existing products with the support of excellent characteristics of cloud.

Because of the Best-effort service in Internet, direct routing path may not always meet the VoIP quality requirements. Thus, most VoIP systems such as SIP+P2P system use application-layer routing scheme to achieve better QoS performance, which is to use relay node to relay RTP stream from the source node to the destination node. The key of this scheme is how to lookup the appropriate relay nodes, which we call relay lookup problem. This paper presents a novel peer relay lookup scheme based on SIP+P2P system. The main idea is to use topology-aware to optimize relay selection. We introduce the mechanism in detail, and then evaluate this mechanism in NS2 network simulation environment. The results show that our scheme is scalable and can get high relay hit ratio, which confirm the feasibility of a real system. We also make comparison with traditional schemes and the results show that our scheme has good path quality.

The Current Internet provides a Best-effort service, which can not satisfy service quality guarantees that VoIP systems often require. Systems such as multi-homing and overlay network leverage path diversity that Internet inherently has to enhance end-to-end application performance. In this paper, we propose a P2P overlay approach, which is used to transmit media stream in SIP+P2P system. SIP+P2P is a Peer-to-Peer VoIP system, which leverages potential path diversity provided by P2P overlay to enhance VoIP quality. Firstly, we propose a path selection mechanism to ensure that end-to-end paths have a high degree of diversity. Then we develop an effective path switching mechanism based on P2P overlay, including path quality evaluation, predicting, and application-driven path switching. Finally, we evaluate our design on our SIP+P2P testbed, and prove that our path selection can explore high path diversity and that path switching can improve VoIP quality effectively.

Multimedia conferencing is becoming a hot topic of communication in recent years. There are already a few products of multimedia conferencing based on H.323. SIP, which is a more feasible protocol, is put on the agenda of being the call signaling protocol for conferencing. Most of the researches on SIP-based multimedia conferencing, however, have still remained on theories or experiments. In this paper, we propose a feasible framework of SIP-based centralized multimedia conferencing, which meets the requirements of the standards and also develops the theories of XCON framework proposed by IETF. We also present an actual implementation of the centralized conferencing server by exploiting open source achievements, using a few accessory devices for multimedia and data collaboration applications which are also implemented in our laboratory. This paper introduces the whole architecture of the practical system, and expatiates on the flows of conference process.

In this paper we present a uniform negotiation and delivery mechanism for SIP-based conferencing system over IP network. With this mechanism rich innovative collaboration services can be developed conveniently to provide users with novel communication experience in IETF centralized conferencing (XCON) system. The mechanism adopts the extended SDP and offer/answer model as media description and negotiation mechanism. And it uses RTP/RTCP and extended MSRP as the media transport control protocols respectively for audio/video and various TCP-based data collaboration applications. In this paper, the requirement of the mechanism is revealed from a high-level design perspective. Then it is analyzed in detail by highlighting several prominent design choices.

SIP-based systems inherently have NAT traversal and QoS problems because of the limited IPV4 addresses and the best-effort service on the Internet. Using relay server to relay media stream can solve NAT traversal in SIP systems, but have less scalability. In this paper, we propose a SIP+P2P architecture for SIP-based systems to solve the above problems. In this novel architecture, light weighted SIP signaling is still client/server based, but media stream is transmitted through P2P network. We embed relay service into some SIP UAs who can be peers, design a P2P overlay network to organize those peer nodes, and provide a fast and efficient relay nodes retrieving mechanism. We also present a detouring technology to select multi good performance paths in P2P overlay and switch real transmitting path between them according to the path performance. We implemented the whole design, and preliminary experiments show that our approach enhanced path reliability and QoS performance of end-to-end communications significantly.

In this paper we present a uniform negotiation and delivery framework for TCP-based data collaboration applications over IP network. With this framework rich innovative collaboration services can be developed conveniently and provide users with novel communication experience. The framework adopts the offer/answer model with extended Session Description Protocol as session description and negotiation mechanism and adopts MSRP as the media transport control protocol. In this paper, the requirement of the framework was revealed from a high-level design perspective and analyzed in detail by highlighting several prominent design choices such as uniform media description and negotiation, MSRP relay selection and TCP connection setup.