An access network for Voice over IP (VoIP) clients (e.g. DOCSIS-based HFC network) often provides a privacy service. However, such a privacy service is limited only to that access network. When VoIP packets are carried over an open IP network or over a network with some connections to the Internet, it is desirable to provide an end-to-end privacy service where each VoIP packet is encrypted at the source and decrypted at the terminating endpoint. Clearly, public key encryption cannot be applied to each voice packet; the performance would be unacceptable regardless of the choice of a public key algorithm. The only alternative is for the two VoIP endpoints to negotiate a shared symmetric key. Since VoIP connections are established only for duration of a phone call, the end-to-end key negotiation needs to occur during each call setup. And it should not noticeably delay the call setup phase. In order to provide end-to-end privacy, it is not sufficient to encrypt all messages between the two endpoints. It is important that the two endpoints authenticate each other - validate each other's identity. Without authentication an adversary might trick two VoIP clients to negotiate keys with her and then sit in the middle of their conversation and record each VoIP packet, before forwarding it to the intended destination. However, direct authentication of the two VoIP endpoints is not always possible in telephony networks - in particular when caller ID blocking services are enabled. To support such anonymity services, it may be sufficient to authenticate not the identity of the caller but the fact that it is a valid subscriber and that all subsequent signaling and voice traffic will be coming from the same source. The PacketCable specifications provide an example of a VoIP architecture with end-to-end privacy that meets the above stated criteria. This paper describes the PacketCable end-to-end privacy approach and suggests additional mechanisms that may be used to further strengthen VoIP privacy under the PacketCable architecture.
This paper presents two approaches to efficient service development for Internet Telephony. In first approach we consider services ranging from core call signaling features and media control as stated in ITU-T's H.323 to end user services that supports user interaction. The second approach supports IETF's SIP protocol. We compare these from differing architectural perspectives, economy of network and terminal development, and propose efficient architecture models for both protocols. In their design, the main criteria were component independence, lightweight operation and portability in heterogeneous end-to-end environments. In proposed architecture, the vertical division of call signaling and streaming media control logic allows for using the components either individually or combined, depending on the level of functionality required by an application.
Interfacing VoIP into the current Public Switched Telephone Network is for the most part done through the use of a gateway. This gateway whether H.323, Session Initiation Protocol or Media Gateway Control Protocol is required for access into a traditional switched telephony network.
Voice over IP (VoIP) is one of the advanced services supported by the next generation mobile communication. VoIP should support various media formats and terminals existing together. This heterogeneous environment may prevent diverse users from establishing VoIP sessions among them. To solve the problem an efficient media negotiation mechanism is required. In this paper, we propose the efficient media negotiation architecture using the transformation server and the Intelligent Location Server (ILS). The transformation server is an extended Session Initiation Protocol (SIP) proxy server. It can modify an unacceptable session INVITE message into an acceptable one using the ILS. The ILS is a directory server based on the Lightweight Directory Access Protocol (LDAP) that keeps user¡*s location information and available media information. The proposed architecture can eliminate an unnecessary response and re-INVITE messages of the standard SIP architecture. It takes only 1.5 round trip times to negotiate two different media types while the standard media negotiation mechanism takes 2.5 round trip times. The extra processing time in message handling is negligible in comparison to the reduced round trip time. The experimental results show that the session setup time in the proposed architecture is less than the setup time in the standard SIP. These results verify that the proposed media negotiation mechanism is more efficient in solving diversity problems.
This paper describers the results and lessons from the voice over IP trial service on the Korea Telecom's VoIP Testbed. The testbed was made up of four different vendors' systems and solutions constituted four separate zones. Even though the backbone network of the testbed was not commercial IP network, we could comprehend some engineering parameters essential to packetized voice QoS. And we got some know-how. These kinds of results will be much help to traditional telco confronted with many difficult issues especially on packet voice network.
Recently, the field of IP Telephony has been experienced considerable evolution through the specification of new protocols and introduction of products implementing these protocols. We visualize IP Telephony evolving to soon offer multiservices encompassing not only voice, but also data, video and multimedia. While the progress has focused on refining protocols and architectures, very little attention has been given to business models for offering these services. This paper introduces the concept of a Service Zone, which from a service provider/network operator perspective fits within the operator's administrative domain, but is viewed as an independent zone with its own management and services, requiring minimal integration with the core network services. Besides its own management, the Enhanced Services Zone may also provide provisioning and maintenance features needed to provide the customer services and availability that subscribers expect from a telephony service providers. The platform must provide reliable service over time, be scalable to meet increased capacity demands, and be upgradeable to incorporate advanced services and features as they become available. Signaling flows are illustrated using SIP and H.323.
The VoIP technology is growing up rapidly, it has big network impact on PT Telkom Indonesia, the bigger telecommunication operator in Indonesia. Telkom has adopted VoIP and one other technology, Intelligent Network (IN). We develop those technologies together in one service product, called Internet Prepaid Calling Card (IPCC). IPCC is becoming new breakthrough for the Indonesia telecommunication services especially on VoIP and Prepaid Calling Card solutions. Network architecture of Indonesia telecommunication consists of three layer, Local, Tandem and Trunck Exchange layer. Network development researches for IPCC architecture are focus on network overlay hierarchy, Internet and PSTN. With this design hierarchy the goal of Interworking PSTN, VoIP and IN calling card, become reality. Overlay design for IPCC is not on Trunck Exchange, this is the new architecture, these overlay on Tandem and Local Exchange, to make the faster call processing. The nodes added: Gateway (GW) and Card Management Center (CMC) The GW do interfacing between PSTN and Internet Network used ISDN-PRA and Ethernet. The other functions are making bridge on circuit (PSTN) with packet (VoIP) based and real time billing process. The CMC used for data storage, pin validation, report activation, tariff system, directory number and all the administration transaction. With two nodes added the IPCC service offered to the market.
The popularity of instant messaging highlights the power that the addition of presence information can bring to communications. Instant messaging systems combine multi-party communications with active presence notifications, allowing users to monitor the presence status of others. We describe several ways presence information can enhance next generation telephone communications and how integration can actually improve instant messaging as well. In addition, we will describe some of the issues associated with implementing and deploying such services, including privacy, data ambiguity and inter-system compatibility.
We present a scalable architecture for assuring Quality of Service to VoIP applications in an Internet Service Provider's network. This architecture is based on the Differentiated Services and Bandwidth Broker models, and can also be used by other resource-sensitive applications. In this paper, we elaborate on a number of significant issues involved in the design, implementation, deployment and use of the Bandwidth Broker. The Call Agent architecture is used as the VoIP application. We describe the Bandwidth Broker prototype that is used to validate our approach. Our findings suggest that it is feasible to use the Bandwidth Broker architecture for assuring QoS, and a trade-off exists between the granularity of resource requests and call-setup delay.
The objective of this paper is to explain QOS usage on PacketCableTM, a cable-industry project aimed at identifying, qualifying, and supporting internet protocol based voice and video products over cable systems. PacketCable uses the Data Over Cable Service Interface Specification (DOCSIS) another cable industry initiative as the underlying platform in its overall architecture. Traditional cable networks are built to provide one-way broadcast services with audio-visual content and limited data. In this present model, the extent to which quality-of- service (QOS) can be used is very restricted. The industry trend is migrating towards providing two-way interactive audio, video, and data services over the cable network. Therefore QOS is becoming an instrumental and inseparable aspect of delivering superior services to consumers. Since PacketCable uses the underlying DOCSIS QOS parameters to establish QOS flows, this paper discusses the linkages between these two protocols.
Though the necessity and importance of quality guarantees for voice over IP networks (VoIP) are well understood and studied, not much has been done to evaluate and measure the quality of service (QoS) for VoIP in a practical commercial environment. This study focuses on an evaluation of media gateway performance, in terms of voice quality, affected by impairments of an IP network in a practical environment. To study critical elements that affect voice quality in general, two end-to-end VoIP networks were built in the Lucent Technologies, Next Generation Network (NGN) Interoperability Lab. Various IP network impairments, such as IP network delay, jitter, and packet loss were introduced into these systems for assessing the IP network impact on voice quality. The performance metric is the end-to-end voice quality. This paper presents the end-to-end VoIP test-bed architecture, the test configuration, the experiment methodology, testing tools, analytical results, and testing results. The performance results are viewed from a user's perspectives in terms of perceptual speech quality measure (PSQM) and speech latency. Finally, the paper points out the crucial factors that affect a successful VoIP network. Some possible remedies are suggested.
Traditionally, speech coding for communication purposes and perceptual audio coding have been separate worlds. On one hand, speech coders provide acceptable speech quality at very low data rates and low delays which are suitable for two-way communication applications, such as Voice over IP (VoIP) or teleconferencing. Due to the underlying coding paradigm, however, such coders do not perform well for non-speech signals (e.g.~music and environmental noise). Furthermore, the sound quality and naturalness is severely limited by the fact that most coders are working in narrow-band mode, i.e. with a bandwidth below 4 kHz. On the other hand, perceptual audio codecs provide excellent subjective audio quality for a broad range of signals including speech at bit rates down to 16 kbit/s. The delay of such a coder/decoder chain, however, usually exceeds 200 ms at very low data rates and in this way is not acceptable for interactive two-way communication. This paper describes a coding scheme which is designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. The codec was standardized within MPEG-4 Version 2 Audio under the work item ``Low Delay Audio Coding'' and is derived from the ISO/MPEG-2/4 Advanced Audio Coding (AAC) algorithm. The algorithm provides modes operating at algorithmic delay as low as 20 ms and is equipped to handle all full-bandwidth high-quality audio signals, both in monophonic, stereophonic and even multi-channel format. Despite of the low algorithmic delay, the codec delivers better audio quality than MPEG-1 Layer-3 (MP3) at the same bit rate. The paper also addresses issues pertaining to the integration of the coder into H.32x and SDP applications.
From the speech quality point of view the differentiation between terminals and network in communications over IP is no longer possible. Consequently the overall speech quality assessment has to take this into account and requires end-to-end tests. Suitable test setups including the terminals acoustics using artificial head technology as a close to reality interface are introduced. In a second part the influence of various subjectively relevant parameters on speech quality is discussed. Correlated objec-tive parameters like delay, echo, double talk capability, listening speech quality and parameters determining background noise transmission quality are described. Appropriate analysis methods are given. The discussion points out the influence of delay on conversation dynamics impairments and its influence on echo perception, because the expected delay in VoIP sce-narios is probably higher than typically recommended for telephone conversations. Optimization criteria are introduced for implemented echo cancellers as well as test methods to assess the one-way speech sound quality, double talk performance and background noise transmission.
Measuring voice quality for telephony is not a new problem. However, packet-switched, best-effort networks such as the Internet present significant new challenges for the delivery of real-time voice traffic. Unlike the circuit-switched PSTN, Internet protocol (IP) networks guarantee neither sufficient bandwidth for the voice traffic nor a constant, minimal delay. Dropped packets and varying delays introduce distortions not found in traditional telephony. In addition, if a low bitrate codec is used in voice over IP (VoIP) to achieve a high compression ratio, the original waveform can be significantly distorted. These new potential sources of signal distortion present significant challenges for objectively measuring speech quality. Measurement techniques designed for the PSTN may not perform well in VoIP environments. Our objective is to find a speech quality metric that accurately predicts subjective human perception under the conditions present in VoIP systems. To do this, we compared three types of measures: perceptually weighted distortion measures such as enhanced modified Bark spectral distance (EMBSD) and measuring normalizing blocks (MNB), word-error rates of continuous speech recognizers, and the ITU E-model. We tested the performance of these measures under conditions typical of a VoIP system. We found that the E-model had the highest correlation with mean opinion scores (MOS). The E-model is well-suited for online monitoring because it does not use the original (undistorted) signal to compute its quality metric and because it is computationally simple.
Special purpose hardware and application software have been developed to implement and test Voice over IP protocols. The hardware has interface units to which ISDN telephone sets can be connected. It has Ethernet and RS-232 interfaces for connections to LANs and controlling PCs. The software has modules which are specific to telephone operations and simulation activities. The simulator acts as a WAN environment, generating delays in delivering speech packets according to delay distribution specified. By using WAN simulator, different algorithms can be tested and their performances can be compared. The novel algorithm developed correlates silence periods with received voice packets and delays play out until confidence is established that a significant phrase or sentence is stored in the playout buffer. The performance of this approach has been found to be either superior or comparable to performances of existing algorithms tested. This new algorithm has the advantage that at least a complete phrase or sentence is played out, thereby increasing the intelligibility considerably. The penalty of having larger delays compared to published algorithms operating under bursty traffic conditions is compensated by higher quality of service offered. In the paper, details of developed system and obtained test results will be presented.
In this paper, the protocol performance of the Voice over IP (VoIP) packet service on the WCDMA radio transport channels is studied. In a packet network, the transport channels between the mobile stations and the radio access network may carry radio bearers with different delay, throughput and quality of service requirements. Voice transmission over the packet network has two key characteristics. First, the quality of service for VoIP needs strict delay requirements for every packet, and the delay variation of the packets should not be extensively high. Secondly, the packet overhead is very large, because of the IP and UDP network headers and the RTP protocol headers present in every packet. This overhead is crucial especially for radio transmission as every bit is known to consume the scarce radio resources. A header compression scheme will significantly reduce the amount of overhead as the static fields of the network headers are needed only at the call setup time to form the context for the compressor-decompressor operation. During the call, it is necessary to send only the difference information of the changing fields, like the RTP timestamp. This enables significant compression ratios to yield headers of mean size less than two to three bytes. In this paper, we have analysed the capabilities to transport VoIP packets on a circuit-switched radio bearer and on a packet-switched radio bearer carried on the dedicated transport channel in a WCDMA cell. A special study was carried out to find out, whether the downlink transport is more efficient in the downlink shared channel compared to the dedicated channel. The analysis is represented in terms of delay, throughput and packet error ratio.
The performance of unreliable voice transmission (Voice over IP) over wireless links is measured not by the throughput but by the perceptual speech quality. The speech quality is impaired by packet losses, which are common on wireless links, and by high transmission delays. In this paper, we describe the design and implementation of a novel Speech Property Based Booster that improves the quality of voice over wireless LANs. This booster is in compliance with existing standards and is transparent to other protocols. It uses characteristics of human speech production and features of modern audio codecs to distinguish packets regarding their importance for perceptual quality. Important packets are protected at the link layer by three mechanisms: selective packet loss recovery, redundant transmission and a hybrid solution. These mechanisms have been evaluated using an experimental set-up with commercial wireless LAN equipment. We made measurements of the objective audio quality and analyzed the effects of packet losses, both due to real wireless channels and late packet arrivals. Our experiments show that the booster increases the quality of voice best with the hybrid solution and that the performance of Voice over IP can be improved further.
The economic advantages of packet voice are driving both the access and core voice networks away from circuit switching towards packet. The industry continues to debate whether the future of these packet networks will be based on pure ATM, pure Internet protocol (IP), IP over asynchronous transfer mode (ATM), IP over multiprotocol label switching, or a combination thereof. There are advantages to both ATM and IP and reasons for choosing each. This paper explores the role of next-generation switches which, as they become widely adopted for both access and core networking, must be able to handle voice traffic over both IP and ATM networks for future extensibility as the debate continues and must have the features necessary to interwork with existing public switched telephone network.