Using Transcoding for Hidden Communication in IP Telephony

The paper presents a new steganographic method for IP telephony called TranSteg (Transcoding Steganography). Typically, in steganographic communication it is advised for covert data to be compressed in order to limit its size. In TranSteg it is the overt data that is compressed to make space for the steganogram. The main innovation of TranSteg is to, for a chosen voice stream, find a codec that will result in a similar voice quality but smaller voice payload size than the originally selected. Then, the voice stream is transcoded. At this step the original voice payload size is intentionally unaltered and the change of the codec is not indicated. Instead, after placing the transcoded voice payload, the remaining free space is filled with hidden data. TranSteg proof of concept implementation was designed and developed. The obtained experimental results are enclosed in this paper. They prove that the proposed method is feasible and offers a high steganographic bandwidth. TranSteg detection is difficult to perform when performing inspection in a single network localisation.Comment: 17 pages, 16 figures, 4 table

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

Peer-to-Peer Communication between Android Mobiles

In the current days Voice telephony over mobile is possible considering GSM system which is cost consuming. Wi-Fi technology is a form of telecommunication that allows data and voice transmissions over a wide range of interconnected networks. In this thesis we provides a mechanism for live communication over IP using mobile phones at no cost. The purpose of this research is to design and implement a telephony program that uses WI-FI in p2p (Peer-to-Peer) or WLAN (Wireless Local Area Network) as a means of communication between mobile phones. The system will allow users to establish p2p voice connection through Access Points (AP) and then allow user to make voice conversation, sending SMS (Short Message Service). The current system will only allow for one call per connection, and no call waiting, or conference calls. The group chat application is the number of users connected to the server

A New covert channel over RTP

In this thesis, we designed and implemented a new covert channel over the RTP protocol. The covert channel modifies the timestamp value in the RTP header to send its secret messages. The high frequency of RTP packets allows for a high bitrate covert channel, theoretically up to 350 bps. The broad use of RTP for multimedia applications, including VoIP, provides plentiful opportunities to use this channel. By using the RTP header, many of the challenges present for covert channels using the RTP payload are avoided. Using the reference implementation of this covert channel, bitrates of up to 325 bps were observed. Speed decreases on less reliable networks, though message delivery was flawless with up to 1% RTP packet loss. The channel is very difficult to detect due to expected variations in the timestamp field and the flexible nature of RTP

Quality of service for VoIP in wireless communications

Ever since telephone services were available to the public, technologies have evolved to more efficient methods of handling phone calls. Originally circuit switched networks were a breakthrough for voice services, but today most technologies have adopted packet switched networks, improving efficiency at a cost of Quality of Service (QoS). A good example of packet switched network is the Internet, a resource created to handle data over an Internet Protocol (IP) that can handle voice services, known as the Voice over the Internet Protocol (VoIP). The combination of wireless networks and free VoIP services is very popular, however its limitations in security and network overload are still a handicap for most practical applications. This thesis investigates network performance under VoIP sessions. The aim is to compare the performance of a variety of audio codecs that diminishes the impact of VoIP in the network. Therefore the contribution of this research is twofold: To study and analyse the extension of speech quality predictors by a new speech quality model to accurately estimate whether the network can handle a VoIP session or not and to implement a new application of network coding for VoIP to increase throughput. The analysis and study of speech quality predictors is based on the mathematical model developed by the E-model. A case study of an embedded Session Initiation Protocol (SIP) proxy, merged with a Media Gateway that bridges mobile networks to wired networks has been developed to understand its effects on QoS. Experimental speech quality measurements under wired and wireless scenarios were compared with the mathematical speech predictor resulting in an extended mathematical solution of the E-model. A new speech quality model for cascaded networks was designed and implemented out of this research. Provided that each channel is modelled by a Markov Chain packet loss model the methodology can predict expected speech quality and inform the QoS manager to take action. From a data rate perspective a VoIP session has a very specific characteristic; exchanged data between two end nodes is often symmetrical. This opens up a new opportunity for centralised VoIP sessions where network coding techniques can be applied to increase throughput performance at the channel. An application layer has been implemented based on network coding, fully compatible with existing protocols and successfully achieves the network capacity.EThOS - Electronic Theses Online ServiceGBUnited Kingdo

Videopuhelun tallentaminen ja toisto

Recording a video call recording is beneficial in cases like job interviews and business meetings. Skype or Google Hangouts for example have no recording implemented without additional plugins. MP4 file container is capable of containing different types of media tracks, is widely used and is supported by media players and has a feature called RTP/RTCP Reception Hint Tracks. Hint tracks contain media transmission instructions, which can be used, e.g., to record RTP stream into a file. Without this information the video call session cannot be replayed afterwards. The purpose of this Thesis is to implement and verify the usage of RTP and RTCP Reception Hint Tracks in video call recording. No open-source MP4 multiplexing library or a media player with support for RTP/RTCP Reception Hint Tracks was found, so the support had to be implemented in both the library and the media player. The setup includes Linphone and L-SMASH for recording and VLC media player for playback. The created MP4 file has two RTP Reception Hint Tracks, two RTCP Reception Hint Tracks, and two media tracks. The GSM audio is chosen because it is supported by Linphone, L-SMASH, and VLC media player. H.264/AVC is chosen for video, because it is the best available codec supported by the three software. Tests were carried out using two laptops with both having recording enabled. From the tests it is concluded that using the RTP Reception Hint Track increases the total CPU usage by less than 1% and the size of the recorded video call by 4% over the conventional media tracks. The implementation shows that RTP Reception Hint Tracks meet well the needs of implementations with choice of different codecs

Video Quality Measurement For 3G Handsets

Internet provides many services. VOIP (Voice over IP) is one such service also known as Internet Telephony or IP Telephony. Using VOIP we can make voice telephony calls, participate in video conferences, etc over data networks (WAN'S and LAN'S) or internet. VOIP operates by first converting voice data into digital form, organizing them into packets, transmitting them through the most convenient route to their destination and finally reassembling them at the destination. Protocols like SIP/RTP, H.323, MGCP are designed which perform all the above steps. This project aims to make a video call from a 3G Mobile to an IP phone via Asterisk Gateway. Asterisk to act as bridge for video call between 3G-IP network must capture the audio/video stream from 3G mobile, convert captured stream into an IP compatible stream and send stream to an IP client and vice-versa. Asterisk needs to support AMR codec for audio and MPEG-4 codec for video and H.324M protocol stack for capturing audio/video streams from 3G Mobile. Asterisk currently supports audio codec's like GSM, G.729, A-law, and U-law. It allows H.261, H.263 video streams as pass-through. It supports VOIP protocols like SIP/RTP, MGCP, and H.323 which allows it to interface with other devices. This project aims to implement AMR codec, H.324M protocol stack, MPEG-4, bridging functions between SIP/RTP-ISDN and 3G Mobile in Asterisk which allows a 3G phone to call a SIP client via Asterisk. This thesis discusses the implementation of AMR in asterisk as well as SIP protocol and SIP soft phones

Micro protocol engineering for unstructured carriers: On the embedding of steganographic control protocols into audio transmissions

Network steganography conceals the transfer of sensitive information within unobtrusive data in computer networks. So-called micro protocols are communication protocols placed within the payload of a network steganographic transfer. They enrich this transfer with features such as reliability, dynamic overlay routing, or performance optimization --- just to mention a few. We present different design approaches for the embedding of hidden channels with micro protocols in digitized audio signals under consideration of different requirements. On the basis of experimental results, our design approaches are compared, and introduced into a protocol engineering approach for micro protocols.Comment: 20 pages, 7 figures, 4 table

Performance evaluation of speech quality for VOIP on the internet

Master'sMASTER OF SCIENC

VOIP WITH ADAPTIVE RATE IN MULTI- TRANSMISSION RATE WIRELESS LANS

“Voice over Internet Protocol (VoIP)” is a popular communication technology that plays a vital role in term of cost reduction and flexibility. However, like any emerging technology, there are still some issues with VoIP, namely providing good Quality of Service (QoS), capacity consideration and providing security. This study focuses on the QoS issue of VoIP, specifically in “Wireless Local Area Networks (WLAN)”. IEEE 802.11 is the most popular standard of wireless LANs and it offers different transmission rates for wireless channels. Different transmission rates are associated with varying available bandwidth that shall influence the transmission of VoIP traffic