Fast RTP Detection and Codecs Classification in Internet Traffic

This paper presents a fast multi-stage method for on-line detection of RTP streams and codec identification of transmitted voice or video traffic. The method includes an RTP detector that filters packets based on specific values from UDP and RTP headers. When an RTP stream is successfully detected, codec identification is applied using codec feature sets. The paper shows advantages and limitations of the method and its comparison with other approaches. The method was implemented as a part of network forensics framework NetFox developed in project SEC6NET. Results show that the method can be successfully used for Lawful Interception as well as for network monitoring

Database of audio records

Diplomka a prakticky castDiplome with partical part

Experimental Evaluation Platform for Voice Transmission Over Internet of Things (VoIoTs)

Internet of Things (IoTs) is an example of the last advances in Information and Communication Technologies. In particular, with the revolutionary development of Wireless Sensor Network (WSN) technologies, researchers largely focused on take benefits of integration embedded low-cost, low-power WSN technology in a various IoTs applications. Real-time voice transmission over IoTs is one interesting application that began to be explored by many researchers. Thus, this paper presents a performance study for transmission of voice over WSN (VoWSN) with and without presence of Internet. A framework using a Raspberry Pi3 (RPi3) and open source FFmpeg technology for processing, compressing and streaming voice to a remote computer is proposed, implemented and evaluated. The performance of the proposed framework is evaluated by studying its behavior utilizing three audio encoding algorithms: AC3, MP3 and OPUS with different sampling rates and a set of evaluation metrics such as :One-way delay, jitter, Bandwidth (B.W), CPU usage and packet losses

Performance Evaluation of a Wireless Network using a VoIP Traffic Generator on a Mobile Device

The problem of generating different patterns of traffic to emulate real user behaviour is receiving considerable attention with the construction of new and more complex network architectures. The theoretical modelling of waveforms or signals that flow through networks is valuable in a variety of scenarios including performance analysis and the design of communication systems. In the literature, many computer-based performance evaluation tools have been discussed. However, these tools lack the ability to run on affordable technologies such as mobile phones. The fundamental contribution of this work is the design of a traffic generating tool called MTGawn which is able to run on a mobile device. Design Science Research was the research methodology used for the design and deployment of a prototype of the proposed system. VoIP traffic was emulated using an implementation of well-known real time transport protocols such as RTP and cRTP, and parameterization was defined by using three codecs namely: G.711, G.723, and G.729. An evaluation was performed in a laboratory wireless network testbed and preliminary results were collected and analysed. The results of the experiments show that such a measuring instrument can be deployed on a mobile phone. More experiments are being done to ensure the accuracy of the data and also to compare the results with that of computer-based systems. Furthermore additional functionalities, similar to the functionality found on the computer-based open source tools, are being added to the mobile tool.Telkom, Cisco, Aria Technologies, THRIPDepartment of HE and Training approved lis

Using the Java Media Framework to build Adaptive Groupware Applications

Realtime audio and video conferencing has not yet been satisfactorily integrated into web-based groupware environments. Conferencing tools are at best only loosely linked to other parts of a shared working environment, and this is in part due to their implications for resource allocation and management. The Java Media Framework offers a promising means of redressing this situation. This paper describes an architecture for integrating the management of video and audio conferences into the resource allocation mechanism of an existing web-based groupware framework. The issue of adaptation is discussed and a means of initialising multimedia session parameters based on predicted QoS is described

A Tool for VoIP Audio Extraction

Cílem práce je vytvořit systém, který dokáže rekonstruovat audio data z VoIP komunikace. Systém rozpozná v záznamu síťového provozu proudy VoIP  paketů a na základě jejich obsahu sestaví přenášený audio signál.  Kromě rozšířeného RTP protokolu je podporován také IAX protokol používaný Asterisk ústřednou, který nabízí zajímavé možnosti a není plně či vůbec podporován dostupnými nástroji. Systém je implementován jako knihovna s minimálním rozhraním.In this thesis, we describe VoIP protocols and design of a system to reconstruct audio data from VoIP communication. The system is able to detect VoIP packet streams in an IP network traffic and assemble an audio signal they carry. RTP and IAX VoIP protocols are supported. Unlike widespread RTP protocol, IAX is not fully supported by available tools although it is used by increasingly popular Asterisk communications project and offers interesting features not found in RTP. The system is implemented as a library with minimal frontend.

Notes on the use of RTP for shared workspace applications

The Real-time Transport Protocol, RTP, has become the dominant protocol for streaming audio and video in IP-based environments. A number of proposals have been made which attempt to build on this success and apply RTP for shared workspace applications. We discuss the needs of such applications and the features provided by RTP, with an aim to showing why RTP is not appropriate for such uses

An Analysis of the MOS under Conditions of Delay, Jitter and Packet Loss and an Analysis of the Impact of Introducing Piggybacking and Reed Solomon FEC for VOIP

Voice over IP (VoIP) is a real time application that allows transmitting voice through the Internet network. Recently there has been amazing progress in this field, mainly due to the development of voice codecs that react appropriately under conditions of packet loss, and the improvement of intelligent jitter buffers that perform better under conditions of variable inter packet delay. In addition, there are other factors that indirectly benefited VoIP. Today, computer networks are faster due to the advances in hardware and breakthrough algorithms. As a result, the quality of VoIP calls has improved considerably. However, the quality of VoIP calls under extreme conditions of packet loss still remains a major problem that needs to be addressed for the next generation of VoIP services. This thesis concentrates in making an analysis of the effects that network impairments, such as: delay, jitter, and packet loss have in the quality of VoIP calls and approaches to solve this problem. Finally, we analyze the impact of introducing forward error correction (FEC) Piggybacking and Reed Solomon codes for VoIP. To measure the mean opinion score of VoIP calls we develop an application based on the E-Model, and utilize perceptual evaluation of speech quality (PESQ)