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Voice over IP (VoIP) is a generic name for ser- vices, systems and technology for telephony over an IP network. It is also referred to as Internet telephony and IP (Internet Pro- tocol) telephony. Internet telephone client softwar...
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Voice over IP (VoIP) is a generic name for ser- vices, systems and technology for telephony over an IP network. It is also referred to as Internet telephony and IP (Internet Pro- tocol) telephony. Internet telephone client software attracted at- tention when it first appeared in 1995. Since that, VoIP has rapidly matured into a practical technology, propelled by the popularization and rapid development of the Internet. IP net- work traffic already exceeds telephone network traffic and is ex- pected to further increase several-fold in the next few years.
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The introduction of the IP multimedia subsystem on 3G cellular networks and the integration with other widely deployed wireless networks based on the IEEE 802.11 protocol family require support for both mobility and quality of ser...
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The introduction of the IP multimedia subsystem on 3G cellular networks and the integration with other widely deployed wireless networks based on the IEEE 802.11 protocol family require support for both mobility and quality of service. When mobile systems move across heterogeneous networks, ongoing real-time sessions are affected not only by handoff delay but also by different packet delay and bit-rate. In this paper we propose a cross-layer mechanism that takes into account mobility at different layers of the network stack in order to yield better quality for VoIP, videoconferencing and other real-time applications.We describe our cross-layer architecture, adaptation techniques, a prototype implementation and experimental results.
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VoIP data is transmitted through a transport protocol called user datagram protocol (UDP) which is intrinsically unreliable. The quality of the voice or multimedia trasmitted during a VoIP session is not much affected after a few ...
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VoIP data is transmitted through a transport protocol called user datagram protocol (UDP) which is intrinsically unreliable. The quality of the voice or multimedia trasmitted during a VoIP session is not much affected after a few packet loss. However, if a secret message is embedded inside VoIP packets using any steganographic method, the integrity of the secret message can be undermined due to the packets being lost during transmission. In this paper, we propose a scheme which is capable of enhancing the reliability of any VoIP steganographic method. We first distribute k message bits into k successive RTP packets. Then, parity bits are used for reconstruction of lost bits caused by packet loss. The implementation of our scheme on matrix embedding using binary Hamming codes steganography results in a reasonable reliability, a good speech quality and a very high steganographic bandwidth of 3050 bps.
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In this paper, we present an approach of integrating SIP (Session Initiation Protocol) in converged multimodal/multimedia communication services. An extensible VoIPTeleserver for VoIP in SIP environment is described. It is based o...
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In this paper, we present an approach of integrating SIP (Session Initiation Protocol) in converged multimodal/multimedia communication services. An extensible VoIPTeleserver for VoIP in SIP environment is described. It is based on the concept of dialogue system and Web convergence that separates the channel dependent media resources from the application dependent service creation and hosting environment. It supports XML based service applications for multiple channels including voice, DTMF, IM and chat over IP. The loosely coupled open architecture in our approach is highly extensible. We describe the concept and structure of VoIPTeleServer used in our approach in detail, which interfaces to the VoIP world through SIP signaling and works as a broker between the VoIP SIP environment and MTIP to deliver converged communication services. A prototype of VoIPTeleServer Was implemented, and services and applications based on SIP and MTIP convergence are constructed. Special attention is given to the adverse effect of delay, jitter and packet loss for voice portal services over IP. In particular, case studies of DTMF service in voice portal under adverse channel conditions are performed. The compounding effects of multiple channel impairments to DTMF in voice portal services over IP are characterized. The potential high error rate of the DTMF service indicates that the data redundancy method as proposed in RFC 2198 is needed for DTMF in order to achieve reliable voice portal services over IP.
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The deployment of multimedia over IP (MoIP), and in particular voice over IP services, requires to solve new security issues they introduce, before completely exploiting the great opportunities they offer to telecommunication mark...
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The deployment of multimedia over IP (MoIP), and in particular voice over IP services, requires to solve new security issues they introduce, before completely exploiting the great opportunities they offer to telecommunication market. Furthermore, the implementation of various security measures can cause a marked deterioration in quality of service, which is fundamental to the operation of an MoIP network that meets users' quality expectations. In particular, because of the time-critical nature of MoIP and its low tolerance for disruption and packet loss, many security measures implemented in traditional data networks are simply not applicable in their current form. This paper presents an analysis of the security options of Session Initiation Protocol- (SIP)-based MoIP architecture aimed at evaluating their impact on delay. In particular, each security option is analyzed in terms of clock cycles needed to perform the related operations. This parameter could be used to estimate the delay introduced by the security mechanisms. Moreover the paper proposes a rigorous definition of five security profiles, which provide different levels of security to a MoIP system.
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IP telephony systems are expected to be de- ployed worldwide in the near future because of their potential for integrating the multimedia communication infrastructure over IP networks. Phone-to-phone connection over an IP network ...
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IP telephony systems are expected to be de- ployed worldwide in the near future because of their potential for integrating the multimedia communication infrastructure over IP networks. Phone-to-phone connection over an IP network via IP telephony gateways (IP-GWs) is a key feature of the system. In an IP telephony system, a low-bit-rate voice codec is used to im- prove bandwidth efficiency. However, due to the packet transfer method over the IP network, it is necessary to add packet headers, including IP, UDP, and RTP headers, which increases the header overhead and thus decreases transfer efficiency. Moreover, be- cause there will be large numbers of short voice packets flowing into the IP network, the load on the Internet will increase. We propose voice stream multiplexing between IP-GWs to solve these problems. In this scheme, multiple voice streams are connected between a pair of IP-GWs, enabling multiplexed voice stream transfer. The voice stream multiplexing mechanism can reduce the header overhead as well as decrease the number of voice pack- ets. The voice stream multiplexing we propose is to concatenate RTP packets destined for the same IP-GW at a multiplexing in- terval period into a single UDP packet. The advantage of this method is that no new additional header is required and the cur- rent well-defined H.323 and RTP standards can be applied with minimum changes. We implemented and tested the system. The results show that the proposed method is effective at reducing both the header overhead and the number of packets. In a typi- cal case, the bandwidth is cut by 40% for eight G.723.1-encoded voice streams through header overhead reduction and the num- ber of voice packets is also decreased to 1/8. Furthermore, this method can easily be enhanced to a general RTP packet multi- plexing method that is applicable not only to an IP-GW but also to other RTP multiplexing and de-multiplexing applications.
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Convergence is one of the most frequently used catchwords in the information and com- munication field. While in general the increasing growing together of telecommunication, information and media technology is understood by this ...
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Convergence is one of the most frequently used catchwords in the information and com- munication field. While in general the increasing growing together of telecommunication, information and media technology is understood by this term and much has already been written about it, the term FMIC has existed for only a short time yet in telecommunication, and has not yet appeared much in the literature. FMIC stands for fixed mobile internet con- vergence and means the growing together of fixed networks, mobile networks and the Internet. In this article it will be demonstrated by means of three top existing current services what one Should expect from the so-called FMIC-services.
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A lot of ongoing work is geared towards improving the reliability, performance and QoS characteristics of service provider IP networks. In contrast, we propose novel enterprise-based techniques that exploit the fact that many ente...
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A lot of ongoing work is geared towards improving the reliability, performance and QoS characteristics of service provider IP networks. In contrast, we propose novel enterprise-based techniques that exploit the fact that many enterprises are multi/dual-homed. In a form of "service-aware routing", certain (e.g., VoIP) packets are duplicated (e.g., at one edge router) and sent over multiple service providers. After traversing the service provider networks, only the first-to-arrive packets are kept and the later-arriving copies are discarded. In so doing, the result is not only better protection against node and link failures, and packet losses and errors, but also better QoS performance under normal (fault-free) operation. The packet-duplication process can be policy-based and take into account costs, bandwidth, and priority issues, permitting the system to behave like a simple "smart router" that automatically and continually makes use of the best (lowest-delay) service provider. In this paper, we present the main ideas behind the proposal, along with some initial analytical and experimental/simulation results and insights from a Linux-based implementation.
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Resource reservation protocol (RSVP) is a network-control protocol used to guarantee Quality-of-Service (QoS) requirements for real-time applications such as Voice-over-IP (VoIP) or Video-over-IP (VIP). However, RSVP was designed ...
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Resource reservation protocol (RSVP) is a network-control protocol used to guarantee Quality-of-Service (QoS) requirements for real-time applications such as Voice-over-IP (VoIP) or Video-over-IP (VIP). However, RSVP was designed for end-systems whose IP addresses do not change. Once mobility of an end-system is allowed, the dynamically changing mobile IP address inevitably impacts on RSVP performance. Our study aims to first quantify the significance of this impact, and then propose a modified RSVP mechanism that provides improved performance during handoffs. Our simulations reveal that the deployment of standard RSVP over Mobile IPv6 (MIPv6) does not yield a satisfactory result, particularly in the case of VIP traffic. Fast Handovers for Mobile IPv6 (FMIPv6) was found to be providing the best performance in all tested scenarios, followed by Hierarchical Mobile IPv6 (HMIPv6) with a single exception: during low handoff rates with VoIP traffic, MIPv6 outperformed HMIPv6. We then designed a new RSVP mechanism, and tested it against standard RSVP. We found that the proposed approach provides a significant improvement of 54.1% in the Total Interruption in QoS (TIQoS) when deployed over a MIPv6 wireless network. For HMIPv6, performance depended primarily on the number of hierarchical levels in the network, with no improvement in TIQoS for single-level hierarchy and up to 37% for a 5-level hierarchy. FMIPv6 on the other hand, provided no room for improvement due to pre-handoff signaling and the tunneling mechanism used to ensure a mobile node (MN)'s connectivity during a handoff, regardless of the RSVP mechanism used.
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Although Voice over IP (VoIP) is rapidly being adopted, its security implications are not yet fully understood. Since VoIP calls may traverse untrusted networks, packets should be encrypted to ensure confidentiality. However, we s...
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Although Voice over IP (VoIP) is rapidly being adopted, its security implications are not yet fully understood. Since VoIP calls may traverse untrusted networks, packets should be encrypted to ensure confidentiality. However, we show that it is possible to identify the phrases spoken within encrypted VoIP calls when the audio is encoded using variable bit rate codecs. To do so, we train a hidden Markov model using only knowledge of the phonetic pronunciations of words, such as those provided by a dictionary, and search packet sequences for instances of specified phrases. Our approach does not require examples of the speaker's voice, or even example recordings of the words that make up the target phrase. We evaluate our techniques on a standard speech recognition corpus containing over 2,000 phonetically rich phrases spoken by 630 distinct speakers from across the continental United States. Our results indicate that we can identify phrases within encrypted calls with an average accuracy of 50%, and with accuracy greater than 90% for some phrases. Clearly, such an attack calls into question the efficacy of current VoIP encryption standards. In addition, we examine the impact of various features of the underlying audio on our performance and discuss methods for mitigation.
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