Time delay estimation algoritms for echo cancellation

The following case study describes how to eliminate echo in a VoIP network using delay estimation algorithms. It is known that echo with long transmission delays becomes more noticeable to users. Thus, time delay estimation, as a part of echo cancellation, is an important topic during transmission of voice signals over packetswitching telecommunication systems. An echo delay problem associated with IP-based transport networks is discussed in the following text. The paper introduces the comparative study of time delay estimation algorithm, used for estimation of the true time delay between two speech signals. Experimental results of MATLab simulations that describe the performance of several methods based on cross-correlation, normalized crosscorrelation and generalized cross-correlation are also presented in the paper

Security in Peer-to-Peer SIP VoIP

VoIP (Voice over Internet Protocol) is one of the fastest growing technologies in the world. It is used by people all over the world for communication. But with the growing popularity of internet, security is one of the biggest concerns. It is important that the intruders are not able to sniff the packets that are transmitted over the internet through VoIP. Session Initiation Protocol (SIP) is the most popular and commonly used protocol of VoIP. Now days, companies like Skype are using Peer-to-Peer SIP VoIP for faster and better performance. Through this project I am improving an already existing Peer-to-Peer SIP VoIP called SOSIMPLE P2P VoIP by adding confidentiality in the protocol with the help of public key cryptography

System Identification with Applications in Speech Enhancement

As the increasing popularity of integrating hands-free telephony on mobile portable devices and the rapid development of voice over internet protocol, identification of acoustic systems has become desirable for compensating distortions introduced to speech signals during transmission, and hence enhancing the speech quality. The objective of this research is to develop system identification algorithms for speech enhancement applications including network echo cancellation and speech dereverberation. A supervised adaptive algorithm for sparse system identification is developed for network echo cancellation. Based on the framework of selective-tap updating scheme on the normalized least mean squares algorithm, the MMax and sparse partial update tap-selection strategies are exploited in the frequency domain to achieve fast convergence performance with low computational complexity. Through demonstrating how the sparseness of the network impulse response varies in the transformed domain, the multidelay filtering structure is incorporated to reduce the algorithmic delay. Blind identification of SIMO acoustic systems for speech dereverberation in the presence of common zeros is then investigated. First, the problem of common zeros is defined and extended to include the presence of near-common zeros. Two clustering algorithms are developed to quantify the number of these zeros so as to facilitate the study of their effect on blind system identification and speech dereverberation. To mitigate such effect, two algorithms are developed where the two-stage algorithm based on channel decomposition identifies common and non-common zeros sequentially; and the forced spectral diversity approach combines spectral shaping filters and channel undermodelling for deriving a modified system that leads to an improved dereverberation performance. Additionally, a solution to the scale factor ambiguity problem in subband-based blind system identification is developed, which motivates further research on subbandbased dereverberation techniques. Comprehensive simulations and discussions demonstrate the effectiveness of the aforementioned algorithms. A discussion on possible directions of prospective research on system identification techniques concludes this thesis

Agent based infrastructure for real-time applications

In this paper we propose a new infrastructure for real-time applications. As a preliminary, we describe basic characteristics of the most popular real-time services like VoIP, videoconferencing, live media streaming, and network multiplayer games. We focus on the end-to-end latency, bandwidth and efficient transmission methods. Next, we present our project concepts, infrastructure model, details of implementation and our testing environment which was designed for testing many aspects of real-time services. The system combines mechanisms for ensuring best possible connection quality (QoS), load balance of servers in infrastructure and gives control over the packet routing decisions. Additionally, provided security mechanisms make it a good choice even in the environment where a high security level is required. The system is based on the Peer-to-Peer (P2P) model and data between users is routed over an overlay network, consisting of all participating peers as network nodes. This overlay can by used for application level multicast or live media stream. In the logging process each user is assigned to a specific node (based on his geographic location and nodes load). Because nodes are participating in data transmission, we have control over the data flow route. It is possible to specify the desired route, so, regardless of the external routing protocol, we can avoid paths that are susceptible to eavesdropping. Another feature of the presented system is usage of agents. Each agent acts within the single node. Its main task is to constantly control the quality of transmission. It analyzes such parameters like link bandwidth use, number of lost packets, time interval between each packet etc. The information collected by the agents from all nodes allows to build a dynamic routing table. Every node uses the Dijkstra's algorithm to find the best at the moment route to all other nodes. The routes are constantly modified as a consequence of changes found by agents or updates sent by other nodes. In VoD services agents also analyze popularity of streamed media, which helps build intelligent video cache. To ensure greater security and high reliability of the system, we have provided a reputation mechanism. It is used during bringing up to date the information about possible routes and their quality, given by other nodes. Owing to this solution nodes and routes which are more reliable get higher priority

Holographic Detection and Reduction of Wind Noise

Many devices that include built-in microphone(s) are used in windy situations. Wind noise degrades the quality of audio detected by the microphone(s), causes microphone signal saturation at high wind speeds, causes nonlinear acoustic echo, and reduces the performance of acoustic echo cancellation (AEC). Applications such as voice‐trigger, automatic speech recognition (ASR), and voice over internet protocol (VoIP) communication are negatively impacted by such degradation. This disclosure describes cost‐effective and robust techniques to detect and reduce wind noise. The described techniques deliver optimum removal and detection results by processing the audio signal in a holographic way by dealing with all related domains including time, frequency, and 3D space. This approach can improve the audio detection performance of any device that incorporates the techniques and can thereby improve the user experience of various applications such as voice-trigger, speech recognition, voice communication, event detection, etc. even on devices that have limited computational capability

An Algorithm to Evaluate the Echo Signal and the Voice Quality in VoIP Networks

Voice over the Internet Protocol (VoIP) has been increasingly popular, but reliability and voice quality remain important factors that limit the widespread adoption of VoIP systems. Providing good voice quality is of major importance for the transition from the PSTN to VoIP networks. There are several non-real-time algorithms that estimate the voice quality such as the PESQ and the E-model. In this thesis we propose a real-time fuzzy algorithm to estimate the echo quality component of the voice quality in VoIP networks. Differently from the existing algorithms, the proposed algorithm does not need a reference signal and has low computational complexity. For these reasons, the proposed algorithm can be embedded in every VoIP system of a network to monitor live calls, giving an estimate of the instantaneous voice quality to the network provider