In this research, we present a investigative study on how timing-based traffic analysis attacks can be used for recovery of the speech from a Voice Over Internet Protocol (VOIP) conversation by taking advantage of the reduction or suppression of the generation of traffic whenever the sender detects a voice inactivity period. We use the simple Bayesian classifier and the complex HMM (Hidden Markov Models) classier to evaluate the performance of our attack. Then we describe the usage of acoustic features in our attack to improve the performance. We conclude by presenting a number of problems that need in-depth study in order to be effective in carrying out silence detection based attacks on VOIP systems.

Abstract Voice over IP (VoIP) is one of the fastest growing applications on the Internet. Concurrently, 802.11 Wireless LANs (WLANs) have become ubiquitous in residential, enterprise, campus and public networks. Currently the majority of traffic on WLANs is data traffic but as more people use wireless networks as their primary access medium, a greater portion of traffic will be real-time traffic such as VoIP traffic. Unfortunately 802.11 networks are designed to handle delay-insensitive, bursty traffic and perform poorly for VoIP streams. Experimental and analytical results have shown that a single 802.11b access point operating at the maximum 11 Mbps rate can support only 5 to 10 VoIP connections simultaneously. Intuitively, an 11 Mbps link should support approximately 85 bi-directional 64Kbps (G.711) streams. The reason for this under-utilization lies primarily in the Distributed Coordination Function (DCF) used by 802.11 MAC layer. The problem can be addressed by using the optional Point Coordination Function (PCF). However PCF is not widely implemented in commodity hardware nor likely to be. There is a similar problem with the proposed 802.11e standard for quality of service. To solve these problems we propose Virtual PCF, a legacy-client compatible solution to increase the number of simultaneous VoIP calls. We implement Virtual PCF, a scheme which employs a variety of techniques to improve both uplink and downlink VoIP QoS. This alleviates delays and packet loss due to DCF contention and doubles the number of supported VoIP sessions.

Abstract Voice over IP (VoIP) is one of the fastest growing applications on the Internet. Concurrently, 802.11 Wireless LANs (WLANs) have become ubiquitous in residential, enterprise, campus and public networks. Currently the majority of traffic on WLANs is data traffic but as more people use wireless networks as their primary access medium, a greater portion of traffic will be real-time traffic such as VoIP traffic. Unfortunately 802.11 networks are designed to handle delay-insensitive, bursty traffic and perform poorly for VoIP streams. Experimental and analytical results have shown that a single 802.11b access point operating at the maximum 11 Mbps rate can support only 5 to 10 VoIP connections simultaneously. Intuitively, an 11 Mbps link should support approximately 85 bi-directional 64Kbps (G.711) streams. The reason for this under-utilization lies primarily in the Distributed Coordination Function (DCF) used by 802.11 MAC layer. The problem can be addressed by using the optional Point Coordination Function (PCF). However PCF is not widely implemented in commodity hardware nor likely to be. There is a similar problem with the proposed 802.11e standard for quality of service. To solve these problems we propose Virtual PCF, a legacy-client compatible solution to increase the number of simultaneous VoIP calls. We implement Virtual PCF, a scheme which employs a variety of techniques to improve both uplink and downlink VoIP QoS. This alleviates delays and packet loss due to DCF contention and doubles the number of supported VoIP sessions.

VoIP technology is one of the important applications of the network. In addition to using traditional PSTN telephone, people can choose favorable VoIP telephone because of the Internet popularization and High speed Internet. Conference is one of the functions of the VoIP, but not every server supported it. Such as OpenSIPS server we use in NSYSU, didn¡¦t supported it. But FreeSWITCH Server has it. Therefore, for solving this problem, I combile two Servers. Then OpenSIPS can use the Conference function. In this paper, we discuss how to combile OpenSIPS and FreeSWITCH. Besides, I have designed a WEB page interface to simplify the procedure to use conference. By this interface, the users can use it to set the timer to start conference. When the timer is up, all of the participants will receive a conference call. After picking up the phone, the users can enter conference.

As network bandwidth growing and voice coding enhancing, voice transmitted over the Packet-Based network environment can also have good quality compared to traditional telephone network. Therefore, the IP telephony services having a price advantage could gradually replace the traditional telecommunication services. However, in order to support both the traditional telecommunication services and IP phone services, it requires the devices which are capable of converting analogical telephony information such as voice and fax into packet data suitable for transmitting over IP. We therefore design and implement an IP phone equipment that can make SIP phone calls, and can support the exchange of the analogical traditional telephone voice and the digital voice packets over IP. In the hardware design point of view, we implement an IP phone set device in an embedded development platform using Infineon EASY5120 development kit, which uses a digital signal processor, called Vinetic®-2CPE (part number is PEB3332), to handle voice encoding and decoding, e.g., G.711 u/A law and G.723 compression, and RTP encapsulation and decapsulation. In the software architecture, we choose Linux as our embedded operation system under which there are lots of GNU open source software to feed our need and further to develop our own software components. In the process of this implementation, the software and hardware co-design takes up most of time, and we also face some VoIP application problems, e.g., SIP. We try to build up a VoIP system to figure out and solve these problems. And we hope that this actually applicable VoIP embedded system can be used as a testbed and verifying platform for VoIP applications.

Telecommunication industry in national legal norm must be able to provide lawful interception functions of offenders phone. The traditional PSTN and GSM have had such a system that can provide investigating authorities to monitor telephone and mobile phone users. In the meanwhile, IP telephony must provide the same monitor functions. However, the current SIP-based IP telephony is still unable to provide this monitoring function. In my thesis, I designed and implemented a monitoring system structure over SIP. It can efficiently carry out lawful interception without violating SIP communication. Additionally, it will not cause any overload on server, but will be able to monitor immediately. The recorded data can be played back without any delay and distortion. A database is built up first for those who are monitored. When SIP dialog begins, SIP proxy inspects whether a call must be monitored. If it is the case of monitoring, a duplicate packet flow is delivered to the monitor. The monitor can playback. I believe this implementation can become a platform for further work in the lawful interception.

As the network fundamental infrastructures become mature, broadband network turns into the main stream. Sufficient bandwidth makes many applications, for example, voice over IP (VoIP), become possible. Through IP phone, people only need to pay local Internet service fee, which is relatively more inexpensive, to be able to make long-distance call with remote people. After the basic calling facility is ready, additional VoIP services become more and more important. User will demand for more additional service functions. In this thesis, I propose and implement a voice recording facility based on SIP-based VoIP system. Users can record both caller and callee's voice together in digital way. Furthermore, we use this facility to provide a voice message recording service. When callee does not pickup his/her phone, caller's phone will be redirected to voice message recording server. Caller can record his/her voice message into callee's directory on the voice recording server, and callee can listen to his/her own voice message later.

It is very convenient for human to use Internet for communications. The VoIP service is a good example. Because voice transmission through Internet becomes mature, people can remotely talk with each other by IP phones inexpensively. Thus, VoIP system can be integrated with Public Switched Telephone Network (PSTN) in some groups, organizations, or companies to reduce the cost for communication. Additionally, it also can be another kind of free consultation channel for customers and users. To take advantage of VoIP system, the people who may be inside a company, be the Internet visitors, or be provisional guests can receive services and question answers immediately and freely. However, it needs a lot of procedures to connect to the VoIP system, e.g., to install a softphone which must be booted to register an VoIP server, to even configure IP address, port number, protocol, encryption, and outbound proxy, etc. This configuration sometimes is difficult and it may need to setup every time when customers want use VoIP phone. It is not an easy job for a user, for example, to download the software from Internet, and to read some documents to setup system parameters and then to operate it, especially for those people who are not familiar with computers. Thus, this inconvenience may cause VoIP service not to be easily promoted. Therefore, the purpose of this paper is to solve this problem and make users have a web interface and without worrying about how to setup the system. I combine Web interface with VoIP phone to become a ¡§webphone¡¨. To take advantage of the characteristics of generality and facility of Web, guests can click the button on the Web pages to trigger the VoIP component inside the Web page to connect to the VoIP system, then make communication with other people. At the same time, it can avoid exposing the information of server address, account, password to public and ward off dangerous attacks from Internet.

With the bandwidth growing and voice coding technology enhancing, VoIP(Voice over IP)becomes a very popular application in recent years. Although VoIP has good voice quality and huge price advantage, it can't replace PSTN(Public Switched Telephone Network) completely in the next few years. We therefore hope to be able to solve the connectivity issue between PSTN and VoIP. In this thesis, I design and implement a VoIP Gateway using Infineon EASY 5120 platform which has a powerful CPU with network processor, ADM5120, and has a DSP (Digital Signal Processor), Vinetic-2CPE PEB3332, to handle digital and analog voice signals. The operation method of the gateway is that PSTN analog signals are passed to Vinetic-2CPE from FXO (Foreign eXchange Office) port, and they are transferred into digital signals to be packetized to transit into Internet to another VoIP Client. My VoIP Gateway can solve the interconnection issues between PSTN and VoIP. Furthermore, in order to solve NAT traversal problem of SIP protocol, we propose and implement an integration approach which uses STUN (Simple Traversal of UDP through NAT) to cooperate with RTP Relay server to traverse various types of NAT, especial symmetric NAT. My VoIP Gateway not only can traverse any type of NAT but also can perform self-configuration according to STUN test report in various network environments.

With the development and improvement of network bandwidth, Voice over Internet Protocol (VoIP) technology begins to bloom. VoIP gradually replaces voice over telecom networks because of its low cost. For instance, VoIP technology divides voice data into different packets and transmits over Internet Protocol, so that it does not require setting up a particular path as the telecom networks does. Although there is much VoIP software available in the market, most of them need pre-installation before using. Our web-based VoIP technology allows users to take advantage of the service with either home computers or public computers. Users only need to log on to our website, then our system will provide users the client program via Java Applet with automatic installation. Users only need to close the browser to terminate the service; the program data will not be left on the computer. In order to provide users to use our service in anytime and anywhere, we need to recognize Network Address Translation (NAT) as one of our biggest barriers. Including Session Initiation Protocol (SIP), instant message and voice data packet could be blocked by NAT. So, we propose a complete solution of NAT traversal. Our system, SIP Communicator, provides a communication platform to users by supporting instant message, VoIP and session transferring. Users can log on to our system via any computers that are JRE supported to communicate with other clients.