Admission control by implicit signaling in support of voice over IP over ADSL

Voice over DSL (VoDSL) is a technology that enables the transport of data and multiple voice calls over a single copper-pair. Voice over ATM (VoATM) and Voice over IP (VoIP) are the two main alternatives for carrying voice over DSL. ATM is currently the preferred technology, since it offers the advantage of ATM’s built-in Quality of Service (QoS) mechanisms. IP QoS mechanisms have been maturing only in recent years. However, if VoIP can achieve comparable performance to that of VoATM in the access networks, it would facilitate end-to-end IP telephony and could result in major cost savings. In this paper, we propose a VoIP-based VoDSL architecture that provides QoS guarantees comparable to those offered by ATM in the DSL access network. Our QoS architecture supports Premium and Regular service categories for voice traffic and the Best-Effort service category for data traffic. The Weighted Fair Queuing algorithm is used to schedule voice and data packets for transmission over the bottleneck link. Fragmentation of large data packets reduces the waiting time for voice packets in the link. We also propose a new admission control mechanism called Admission Control by Implicit Signaling. This mechanism takes advantage of application layer signaling by mapping it to the IP header. We evaluate the performance of our QoS architecture by means of a simulation study. Our results show that our VoIP architecture can provide QoS comparable to that provided by the VoATM architecture.     

QoS-aware path switching for VoIP traffic using SCTP

Voice over Internet protocol (VoIP) has been a prevalent multimedia service nowadays. It allows us to transmit voice data over IP networks. However, quality of service (QoS) is a major challenge to VoIP services. It must provide similar quality to traditional public switched telephone network or cellular phone services. Therefore, QoS related protocols have become important for real-time applications. Multi-protocol label switch (MPLS) is one of the important techniques to improve the network performance from QoS point of view. It employs label swapping to speed up packet forwarding. However, when a large number of users utilize VoIP services, the network congestion issue still exists. It causes delay, jitter and packet loss that affect VoIP QoS. In this paper, we propose a QoS-aware path switching strategy by using stream control transmission protocol (SCTP) in MPLS network to improve the VoIP traffic. This was done by employing SCTP selective acknowledgment mechanism to report the transmission parameters of primary path and to determine the criteria to switch to backup path. Simulation results show significant improvement in VoIP QoS.     

Voice-quality monitoring and control for VoIP

Voice over IP offers important opportunities for the telecommunications market to deploy more advanced services, but it must overcome many obstacles. Users expect toll-quality voice, which calls for end-to-end quality of service (QoS) - a challenge for IP service providers. To make VoIP attractive to end users, the only feasible and directly implementable alternative is to deploy an efficient mechanism within the endpoints. To that end, the authors propose the scalable, modular, call quality monitoring and control framework for maintaining voice quality at acceptable levels over networks that don't offer QoS guarantees.     

A multi-path mechanism for reliable VoIP transmission over wireless networks

With the advances in audio encoding standards and wireless access networks, voice over IP (VoIP) is becoming quite popular. However, real-time voice data over lossy networks (such as WLAN and UMTS), still posses several challenging problems because of the adverse effects caused by complex network dynamics. One approach to provide QoS for VoIP applications over the wireless networks is to use multiple paths to deliver VoIP data destined for a particular receiver. This paper introduced cmpSCTP, a transport layer solution for concurrent multi-path transfer that modifies the standard stream control transmission protocol (SCTP). The cmpSCTP aims at exploiting SCTP’s multi-homing capability by selecting several best paths among multiple available network interfaces to improve data transfer rate to the same multi-homed device. Through the use of path monitoring and packet allotment techniques, cmpSCTP tries to transmit an amount of packets corresponding to the path’s ability. At the same time, cmpSCTP updates the transmission strategy based on the real-time information of all of paths. Using cmpSCTP’s flexible path management capability, we may switch the flow between multiple paths automatically to realize seamless path handover. The theoretical analysis evaluated the model of cmpSCTP and formulated optimal traffic fragmentation of VoIP data. Extensive simulations under different scenarios using OPNET verified that cmpSCTP can effectively enhance VoIP transmission efficiency and highlighted the superiority of cmpSCTP against the other SCTP’s extension implementations under performance indexes such as throughput, handover latency, packet delay, and packet loss.     

VoIP中为提高语音质量所采用的关键技术

VoIP(Voice over IP)即IP电话,是将话音编码、压缩转换成数据包,在IP网络中进行传输的技术。为应对传统电话公司的竞争,IP电话的语音质量成为决定其未来命运的关键因素。该文首先介绍了几个界定QoS的参数和目前评价IP电话业务语音质量的三种模型：MOS模型、PSQM模型、E模型,然后重点介绍了在终端和网络上提高VoIP语音质量所采用的一些关键技术,应用于终端的技术中比较重要的是语音的编码与压缩、差错控制等,而应用于网络的技术则是解决IP QoS的两种基本模型：综合业务模型和区分业务模型。     

无线Mesh网络中VOIP的QoS性能分析

针对VoIP(Voice over IP)业务在无线Mesh网上进行传输时存在服务质量(QoS)需求难以保证、带宽利用率低的问题,介绍了VoIP的QoS影响因素,分析了端到端时延、时延抖动和丢包率等几个重要参数,并对VoIP在无线Mesh网中的传输性能进行了论述。提出了基于无线Mesh网络的QoS保证机制,可以为端到端的数据传输公平的分配带宽,并能在保证QoS下实现大规模的实时任务的多跳转发。仿真试验表明能有效降低端到端时延,有着更好的QoS性能。     

Analysis of Skype VoIP traffic in UMTS: End-to-end QoS and QoE measurements

In the future Internet, multi-network services will follow a new paradigm in which the intelligence of the network control is gradually moved to the edge of the network. This impacts both the objective Quality of Service (QoS) of the end-to-end connection as well as the subjective Quality of Experience (QoE) as perceived by the end user. Skype already offers such a multi-network Voice-over-IP (VoIP) telephony service today. Due to its ease of use and a high sound quality, it becomes increasingly popular in the wired Internet.UMTS operators promise to offer large data rates which should suffice to support VoIP calls in a mobile environment. However, the success of those applications strongly depends on the corresponding QoE. In this work, we analyze the theoretically achievable as well as the actually achieved quality of IP-based voice calls using Skype. This is done performing measurements in both a real UMTS network and a testbed environment. The latter is used to emulate rate control mechanisms and changing system conditions of UMTS networks. The results show in how far Skype over UMTS is able to keep pace with existing mobile telephony systems and how it reacts to different network characteristics. The investigated performance measures comprise the QoE in terms of the MOS value and the QoS in terms of network-based factors like throughput, packet interarrival times, or packet loss.     

What are suspicious VoIP delays?

Voice over IP (VoIP) is unquestionably the most popular real-time service in IP networks today. Recent studies have shown that it is also a suitable carrier for information hiding. Hidden communication may pose security concerns as it can lead to confidential information leakage. In VoIP, RTP (Real-time Transport Protocol) in particular, which provides the means for the successful transport of voice packets through IP networks, is suitable for steganographic purposes. It is characterised by a high packet rate compared to other protocols used in IP telephony, resulting in a potentially high steganographic bandwidth. The modification of an RTP packet stream provides many opportunities for hidden communication as the packets may be delayed, reordered or intentionally lost. In this paper, to enable the detection of steganographic exchanges in VoIP, we examined real RTP traffic traces to answer the questions, what do the “normal” delays in RTP packet streams look like? and, is it possible to detect the use of known RTP steganographic methods based on this knowledge?     

Call admission control and load balancing for voice over IP

IP networks are traditionally designed to support a best-effort service, with no guarantees on the reliable and timely delivery of packets. With the migration of real-time applications such as voice onto IP-based platforms, the existing IP network capabilities become inadequate to provide the quality-of-service (QoS) levels that the end-users are accustomed to. While new protocols such as DiffServ and MPLS allow some amount of traffic prioritization, guaranteed QoS requires call admission control. This paper reviews several possible implementations and shows simulation results for one promising method that makes efficient use of the network and is scalable to large networks.     

Evaluating perceived voice quality on packet networks using different random neural network architectures

Voice over Internet Protocol (VoIP) is one of the fastest growing technologies in the world. In VoIP speech signals are transmitted over the same network used for data communications. The internet is not a robust network and is subjected to delay, jitter, and packet loss. It is very important to measure and monitor the quality of service (QoS) the users experience in VoIP networks; this is not an easy task and usually requires subjective tests. In this paper we have analyzed three non-intrusive models to measure and monitor voice quality using Random Neural Networks (RNN). A RNN is an open queuing network with positive and negative signals. We have assessed the voice quality based on various parameters i.e. delay, jitter, packet loss, and codec. In our approach we have used the Mean Opinion Score (MOS) calculated using a Perceptual Evaluation of Speech Quality (PESQ) algorithm to generate data for training the RNN model. We have studied two feed-forward models and a recurrent architecture. We have found that the simple feed-forward architecture has produced the most accurate results compared to the other two architectures.     

An Overlay Architecture for High-Quality VoIP Streams

The cost savings and novel features associated with voice over IP (VoIP) are driving its adoption by service providers. Unfortunately, the Internet's best effort service model provides no quality of service guarantees. Because low latency and jitter are the key requirements for supporting high-quality interactive conversations, VoIP applications use UDP to transfer data, thereby subjecting themselves to quality degradations caused by packet loss and network failures. In this paper, we describe an architecture to improve the performance of such VoIP applications. Two protocols are used for localized packet loss recovery and rapid rerouting in the event of network failures. The protocols are deployed on the nodes of an application-level overlay network and require no changes to the underlying infrastructure. Experimental results indicate that the architecture and protocols can be combined to yield voice quality on par with the public switched telephone network     

An efficient joint channel assignment and QoS routing protocol for IEEE 802.11 multi-radio multi-channel wireless mesh networks

With the emerging of video, voice over IP (VoIP) and other real-time multimedia services, more and more people pay attention to quality of service (QoS) issues in terms of the bandwidth, delay and jitter, etc. As one effective way of broadband wireless access, it has become imperative for wireless mesh networks (WMNs) to provide QoS guarantee. Existing works mostly modify QoS architecture dedicated for ad hoc or sensor networks, and focus on single radio and single channel case. Meanwhile, they study the QoS routing or MAC protocol from view of isolated layer. In this paper, we propose a novel cross-layer QoS-aware routing protocol on OLSR (CLQ-OLSR) to support real-time multimedia communication by efficiently exploiting multi-radio and multi-channel method. By constructing multi-layer virtual logical mapping over physical topology, we implement two sets of routing mechanisms, physical modified OLSR protocol (M-OLSR) and logical routing, to accommodate network traffic. The proposed CLQ-OLSR is based on a distributed bandwidth estimation scheme, implemented at each node for estimating the available bandwidth on each associated channel. By piggybacking the bandwidth information in HELLO and topology control (TC) messages, each node disseminates information of topology and available bandwidth to other nodes in the whole network in an efficient way. From topology and bandwidth information, the optimized path can be identified. Finally, we conduct extensive simulation to verify the performance of CLQ-OLSR in different scenarios on QualNet platform. The results demonstrate that our proposed CLQ-OLSR outperforms single radio OLSR, multi-radio OLSR and OLSR with differentiated services (DiffServ) in terms of network aggregate throughput, end-to-end packet delivery ratio, delay and delay jitter with reasonable message overheads and hardware costs. In particular, the network aggregate throughput for CLQ-OLSR can almost be improved by 300% compared with the single radio case.     

Using cross-layer metrics to improve the performance of end-to-end handover mechanisms

Network centric handover solutions for all IP wireless networks usually require modifications to network infrastructure which can stifle any potential rollout. This has led researchers to begin looking at alternative approaches. Endpoint centric handover solutions do not require network infrastructure modification, thereby alleviating a large barrier to deployment. Current endpoint centric solutions capable of meeting the delay requirements of Voice over Internet Protocol (VoIP) fail to consider the Quality of Service (QoS) that will be achieved after handoff. The main contribution of this paper is to demonstrate that QoS aware handover mechanisms which do not require network support are possible. This work proposes a Stream Control Transmission Protocol (SCTP) based handover solution for VoIP called Endpoint Centric Handover (ECHO). ECHO incorporates cross-layer metrics and the ITU-T E-Model for voice quality assessment to accurately estimate the QoS of candidate handover networks, thus facilitating a more intelligent handoff decision. An experimental testbed was developed to analyse the performance of the ECHO scheme. Results are presented showing both the accuracy of ECHO at estimating the QoS and that the addition of the QoS capabilities significantly improves the handover decisions that are made.     

Application-aware scheduling for VoIP in Wireless Mesh Networks

Wireless Mesh Networks (WMNs) are seen as a means to provide last mile connections in Next Generation Networks (NGNs). Because of their auto-configuration capabilities and the low deployment cost WMNs are considered to be an efficient solution for the support of multiple voice, video and data services in NGNs. This paper looks at the optimal provision of resources in WMNs for Voice over IP (VoIP) traffic, which has strict performance requirements in terms of delay, jitter and packet loss. In WMNs, because of the challenges introduced by wireless multi-hop transmissions and limited resources, providing performance quality for VoIP comparable to the voice quality in the traditional circuit-switched networks is a major challenge.This paper analyses different scheduling mechanisms for TDMA-based access control in mesh networks as specified in the IEEE 802.16-2004 WiMAX standard. The performance of the VoIP applications when different scheduling mechanisms are deployed is analysed on a variety of topologies using ns-2 simulation and mathematical analysis. The paper concludes that on-demand scheduling of VoIP traffic – typically deployed in 802.11-based WMNs – is not able to provide the required VoIP quality in realistic mesh WiMAX network scenarios and is therefore not optimal from a network operator’s point of view. Instead, it is shown, that continuous scheduling is much better suited to serve VoIP traffic. The paper then proposes a new VoIP-aware resource coordination scheme and shows, through simulation, that the new scheme is scalable and provides good quality for VoIP service in a wide range of network scenarios. The results shown in the paper prove that the new scheme is resilient to increasing hop count, increasing number of simultaneous VoIP sessions and the background traffic load in the network. Compared to other resource coordination schemes the VoIP-aware scheduler significantly increases the number of supported calls.     

Providing MAC QoS for multimedia traffic in 802.11e based multi-hop ad hoc wireless networks

Ad hoc wireless networks with their widespread deployment, now need to support applications that generate multimedia and real-time traffic. Video, audio, real-time voice over IP, and other multimedia applications require the network to provide guarantees on the Quality of Service (QoS) of the connection. The 802.11e Medium Access Control (MAC) protocol was proposed with the aim of providing QoS support at the MAC layer. The 802.11e performs well in wireless LANs due to the presence of Access Points (APs), but in ad hoc networks, especially multi-hop ones, it is still incapable of supporting multimedia traffic.One of the most important QoS parameters for multimedia and real-time traffic is delay. Our primary goal is to reduce the end-to-end delay, thereby improving the Packet Delivery Ratio of multimedia traffic, that is, the proportion of packets that reach the destination within the deadline, in 802.11e based multi-hop ad hoc wireless networks.Our contribution is threefold: first we propose dynamic ReAllocative Priority (ReAP) scheme, wherein the priorities of packets in the MAC queues are not fixed, but keep changing dynamically. We use the laxity and the hop length information to decide the priority of the packet. ReAP improves the PDR by over 28% in comparison with 802.11e, especially under heavy loads. Second, we introduce Adaptive-TXOP (A-TXOP), where transmission opportunity (TXOP) is the time interval during which a node has the right to initiate transmissions. This scheme reduces the delay of video traffic by reducing the number of channel accesses required to transmit large video frames. It involves modifying the TXOP interval dynamically based on the packets in the queue, so that fragments of the same packet are sent in the same TXOP interval. A-TXOP is implemented over ReAP to further improve the performance of video traffic. ReAP with A-TXOP helps in reducing the delay of video traffic by over 27% and further improves the quality of video in comparison with ReAP without A-TXOP. Finally, we have TXOP-sharing, which is aimed at reducing the delay of voice traffic. It involves using the TXOP to transmit to multiple receivers, in order to utilize the TXOP interval fully. It reduces the number of contentions to the channel and thereby reduces the delay of voice traffic by over 14%. A-TXOP is implemented over ReAP to further improve the performance of voice traffic. The three schemes (ReAP, A-TXOP, and TXOP-sharing) work together to improve the performance of multimedia traffic in 802.11e based multi-hop ad hoc wireless networks.     

Quality-of-service in packet networks: basic mechanisms and directions

In this paper, we review the basic mechanisms used in packet networks to support Quality-of-Service (QoS) guarantees. We outline the various approaches that have been proposed, and discuss some of the trade-offs they involve. Specifically, the paper starts by introducing the different scheduling and buffer management mechanisms that can be used to provide service differentiation in packet networks. The aim is not to provide an exhaustive review of existing mechanisms, but instead to give the reader a perspective on the range of options available and the associated trade-off between performance, functionality, and complexity. This is then followed by a discussion on the use of such mechanisms to provide specific end-to-end performance guarantees. The emphasis of this second part is on the need for adapting mechanisms to the different environments where they are to be deployed. In particular, fine grain buffer management and scheduling mechanisms may be neither necessary nor cost effective in high speed backbones, where “aggregate” solutions are more appropriate. The paper discusses issues and possible approaches to allow coexistence of different mechanisms in delivering end-to-end guarantees.     

Softwarization and virtualization of VoIP networks

Nowadays, voice over IP (VoIP) is a cost-effective and efficient technology in the communications industry. Free applications for transferring multimedia on the Internet are becoming more attractive and pervasive day by day. Nevertheless, the traditional, close, and hardware-defined nature of the VoIP networks’ structure makes the management of these networks more complicated and costly. Besides, its elementary and straightforward mechanisms for routing call requests have lost their efficiency, causing some problems, such as SIP servers’ overload. In order to tackle these problems, we introduce VoIP network softwarization and virtualization and propose two novel frameworks in this article. In this regard, we take advantage of the SDN and NFV concepts such that we separate data and control planes and provide the possibility for centralized and softwarized control of this network. This matter leads to effective routing. The NFV also makes this network’s dynamic resource management possible by functions virtualization of the VoIP network. The proposed frameworks are implemented in a real testbed, including Open vSwitch and Floodlight, examined by various scenarios. The results demonstrate an improvement in signaling and media quality in the VoIP network. As an example, the average throughput and resource efficiency increased by at least 28% and the average response time decreased by 34%. The overall latency has also been reduced by almost 39%.

     

A cross-layer elastic CAC and holistic opportunistic scheduling for QoS support in WiMAX

QoS provisioning is an important issue in the deployment of broadband wireless access networks with real-time and non-real-time traffic integration. An opportunistic MAC (OMAC) combines cross-layer design features with opportunistic scheduling scheme to achieve high system utilization while providing QoS support to various applications. A single scheduling algorithm cannot guarantee all the QoS requirements of traffics without the support of a suitable CAC and vice versa. In this paper, we propose a cross-layer MAC scheduling framework and a corresponding opportunistic scheduling algorithm in tandem with the CAC algorithm to support QoS in WiMAX point-to-multipoint (PMP) networks. Extensive experimental simulations have been carried out to evaluate the performance of our proposal. The simulation results show that our proposed solution can improve the performance of WiMAX networks in terms of packet delay, packet loss rate and throughput. The proposed CAC scheme can guarantee the admitted connections to meet their QoS requirements.     

Quality of service: old idea, new options

ATM has offered QoS guarantees for nearly a decade, but the push is now for IP-based solutions. IP is ubiquitous in today's congested networks. Applications are more complex, users are more demanding, standards bodies are more receptive, and technology is more sophisticated. All this has focused attention on ways to add QoS to IP networks without exorbitant cost. But with progress has come some confusion. Telecommunications carriers and service providers are already enticing customers by offering two or three distinct classes of service over their IP networks. Vendors are beginning to ship QoS-capable hardware and software. ATM is firmly established in the Internet's core. A lot has happened in a short time, which means that users and providers must be aware of where things are going and what the various QoS technologies can actually do. Some QoS mechanisms deliver strict, absolute performance guarantees. Others merely offer assurances that one service class will take priority over another when resources are scarce