Open communication over the Internet poses a serious threat to countries with repressive regimes, leading them to develop and deploy censorship mechanisms within their networks. Unfortunately, existing censorship circumvention systems face difficulties in providing unobservable communication with their clients; this highly limits their availability as censors can easily block access to circumvention systems that make observable communication patterns. Moreover, the lack of unobservability may pose serious threats to their users. Recent research takes various approaches to tackle this problem, however they introduce new challenges, and the provided unobservability is breakable. In this paper we propose an easy-to-deploy and unobservable censorship-resistant infrastructure, called FreeWave. FreeWave works by modulating a client’s Internet traffic into acoustic signals that are carried over VoIP connections. Such VoIP connections are targeted to a server, the FreeWave server, that extracts the tunneled traffic and proxies them to the uncensored Internet. The use of actual VoIP connections, as opposed to traffic morphing, allows FreeWave to relay its VoIP connections through oblivious VoIP nodes (e.g., Skype supernodes), hence keeping the FreeWave server(s) unobservable and unblockable. In addition, the use of end-to-end encryption, which is supported/mandated by most VoIP providers like Skype, prevents censors from distinguishing FreeWave’s VoIP connections from regular VoIP connections. To utilize a VoIP connection’s throughput efficiently we design communications encoders tailored specifically for VoIP’s lossy channel. We prototype FreeWave over Skype, the most popular VoIP system. We show that FreeWave is able to reliably achieve communication throughputs that are sufficient for web browsing, even when clients are far distanced from the FreeWave server. We also validate FreeWave’s communication unobservability against traffic analysis and standard censorship techniques. Keywords-Internet Censorship; Circumvention; VoIP;

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I want my voice to be heard: IP over Voice-over-IP for unobservable censorship circumvention.

The IETF recently standardized the Opus codec as RFC6716. Opus targets a wide range of real-time Internet applications by combining a linear prediction coder with a transform coder. We describe the transform coder, with particular attention to the psychoacoustic knowledge built into the format. The result out-performs existing audio codecs that do not operate under real-time constraints.

High-Quality, Low-Delay Music Coding in the Opus Codec

In this paper, we describe the voice mode of the Opus speech and audio codec. As only the decoder is standardized, the details in this paper will help anyone who wants to modify the encoder or gain a better understanding of the codec. We go through the main components that constitute the voice part of the codec, provide an overview, give insights, and discuss the design decisions made during the development. Tests have shown that Opus quality is comparable to or better than several state-of-the-art voice codecs, while covering a much broader application area than competing codecs.

Voice Coding with Opus

The present invention relates to a method for processing an audio signal, comprising: a step of performing a frequency conversion process on an audio signal to obtain a plurality of frequency transform coefficients; a step of selecting either a general mode or a non-general mode, on the basis of a pulse ratio, for the frequency transform coefficients having a high frequency band from among the plurality of frequency transform coefficients; and a step of performing, if the non-general mode is selected, the following steps: extracting a predetermined number of pulses from the frequency transform coefficients having the high frequency band, and generating pulse information; generating an original noise signal from the frequency transform coefficients having the high frequency band, excluding the pulses; generating a reference noise signal using the frequency transform coefficient having a low frequency band from among the plurality of frequency transform coefficients; and generating noise position information and noise energy information using the original noise signal and the reference noise signal

Method and apparatus for processing an audio signal

This paper discusses the voice quality of Opus, IETF driven open source voice and audio codec. Opus is a newly developed hybrid codec based on SILK and CELT codec technologies. Opus construction is described shortly in this paper and more importantly its optimal operating points are found out based on the listening test results. Voice quality was evaluated with two subjective listening tests. Industry standard voice codecs: 3GPP AMR and AMR-WB, and ITU-T G.718B, G.722.1C and G.719 as well as direct signals were used as voice quality references.

Voice Quality Characterization of IETF Opus Codec.

This document describes SILK, a speech codec for real-time, packet-
based voice communications. Targeting a diverse range of operating
environments, SILK provides scalability in several dimensions. Four
different sampling frequencies are supported for encoding the audio
input signal. Adaptation to network characteristics is provided
through control of bitrate, packet rate, packet loss resilience and
use of discontinuous transmission (DTX). And several different
complexity levels let SILK take advantage of available processing
power without relying on it. Each of these properties can be adjusted
during operation of the codec on a frame-by-frame basis.

Soeren Jensen

Papers

SILK Speech Codec