Internet Telephony: architecture and protocols – an IETF perspective

Internet telephony offers the opportunity to design a global multimedia communications system that may eventually replace the existing telephony infrastructure. We describe the upper-layer protocol components that are specific to Internet telephony services: the Real-Time Transport Protocol (RTP) to carry voice and video data, and the Session Initiation Protocol (SIP) for signaling. We also mention some complementary protocols, including the Real Time Streaming Protocol (RTSP) for control of streaming media, and the Wide Area Service Discovery Protocol (WASRV) for location of telephony gateways.     

远程教学直播系统中媒体同步的研究与实现

对于远程教学直播系统这样包含不同媒体流的分布式多媒体应用而言，媒体同步是一项重要内容。利用RTP（实时传输协议）传输机制中的时间戳和序列号信息，提出同步控制算法，实现了流内同步和流间同步。考虑到分组网络带来的延时抖动，算法可以动态地适应网络延时变化，从而保证了分布式环境中媒体同步的服务质量。     

Generic forward error correction of short frames for IP streaming applications

If the frame size of a multimedia encoder is small, Internet Protocol (IP) streaming applications need to pack many encoded media frames in each Real-time Transport Protocol (RTP) packet to avoid unnecessary header overhead. The generic forward error correction (FEC) mechanisms proposed in the literature for RTP transmission do not perform optimally in terms of stability when the RTP payload consists of several individual data elements of equal priority. In this paper, we present a novel approach for generating FEC packets optimized for applications packing multiple individually decodable media frames in each RTP payload. In the proposed method, a set of frames and its corresponding FEC data are spread among multiple packets so that the experienced frame loss rate does not vary greatly under different packet loss patterns. We verify the performance improvement gained against traditional generic FEC by analyzing and comparing the variance of the residual frame loss rate in the proposed packetization scheme and in the baseline generic FEC.     

Synchronized delivery and playout of distributed stored multimedia streams

Multimedia streams such as audio and video impose tight temporal constraints for their presentation. Often, related multimedia streams, such as audio and video, must be presented in a synchronized way. We introduce a novel scheme to ensure the continuous and synchronous delivery of distributed stored multimedia streams across a communications network. We propose a new protocol for synchronized playback and compute the buffer required to achieve both, the continuity within a single substream and the synchronization between related substreams. The scheme is very general and does not require synchronized clocks. Using a resynchronization protocol based on buffer level control, the scheme is able to cope with server drop-outs and clock drift. The synchronization scheme has been implemented and the paper concludes with our experimental results.     

Internet多媒体分层组播JSCC差错控制率失真优化

针对Internet多媒体群组通信中同时存在的带宽异构性和包丢失率异构性，文中将分层组播和接收者驱动的思想扩展到FEC差错控制中，提出一种分层FEC组播差错控制方法LM-FEC．LM-FEC通过不同的组播组发送信源编码层和各信源层的FEC校验数据，为接收者根据信道带宽和数据包丢失率实施差错控制提供更加灵活的选择．文中用FH-MDP模型描述接收者行为，通过JSCC率失真优化确定编码层内和编码层间的速率分配，JSCC率失真优化采用变量替换和动态规划算法求解．实验表明，该文提出的差错控制方法能够有效改善重建多媒体信号的回放质量．     

Multiple multicast groups for multimedia on the Internet

The use of large-scale multimedia applications has escalated in recent years. Distributed object frameworks such as the Common Object Request Broker Architecture are being deployed to cater this market. We argue however, that CORBA is lacking for large-scale multimedia applications requiring timely guarantees.The Real-Time Wide Area Network dissemination Architecture is a framework for distributing multimedia using dynamically tailored protocol stacks and multiple multicast groups over the Internet. RWANDA improves the quality of reception within heterogeneous receivers by allowing each receiver to subscribe to a one or more QoS multicast groups according to its resources. It overcomes network congestion and heterogeneity using multiple multimedia multicast groups and the dynamic construction of protocol stocks allow receivers to process messages with no prior knowledge of the senders configuration. Protocol details are presented in this article with experimental and simulation results to back our claims.     

A secure and efficient ECC-based user anonymity-preserving session initiation authentication protocol using smart card

The Session Initiation Protocol (SIP) is a signaling communications protocol, which has been chosen for controlling multimedia communication in 3G mobile networks. The proposed authentication in SIP is HTTP digest based authentication. Recently, Tu et al. presented an improvement of Zhang et al.’s smart card-based authenticated key agreement protocol for SIP. Their scheme efficiently resists password guessing attack. However, in this paper, we analyze the security of Tu et al.’s scheme and demonstrate their scheme is still vulnerable to user’s impersonation attack, server spoofing attack and man-in-the middle attack. We aim to propose an efficient improvement on Tu et al.’s scheme to overcome the weaknesses of their scheme, while retaining the original merits of their scheme. Through the rigorous informal and formal security analysis, we show that our scheme is secure against various known attacks including the attacks found in Tu et al.’s scheme. Furthermore, we simulate our scheme for the formal security analysis using the widely-accepted AVISPA (Automated Validation of Internet Security Protocols and Applications) tool and show that our scheme is secure against passive and active attacks including the replay and man-in-the-middle attacks. Additionally, the proposed scheme is comparable in terms of the communication and computational overheads with Tu et al.’s scheme and other related existing schemes.     

Performance simulation of wireless multimedia systems using NC-PRMA/DA and SNC-PRMA/DA protocols

A noncollision packet reservation multiple access with dynamic allocation (NC-PRMA/DA) scheme is proposed and investigated as a suitable candidate protocol for wireless multimedia communications. Access requests of the existing users in NC-PRMA/DA are conveyed to the base station in a noncollision manner by using a time-frequency signaling scheme. In addition, the scheduling NC-PRMA/DA (SNC-PRMA/DA) scheme, which is NC-PRMA/DA with packet scheduling, is also presented. Mixed voice, variable-rate video, and data traffic is used to demonstrate the efficiency of the proposed schemes for a single-cell environment. Simulation results show that our proposed NC-PRMA/DA and SNC-PRMA/DA protocols outperform the existing PRMA/DA protocol.     

大型动态多播群组的分布式密钥管理方案

多播是一种基于Internet的一对多或多对多的有效通信技术，随着各种大型多播应用的迅速发展，在Internet上提供一个分布式的多播密钥管理协议成了一个亟待解决的重要课题．首先分析了已有的一些典型协议，讨论了它们的优点及其存在的问题，在研究了大型动态多播群组的特点及密钥管理要求的基础上，对这些协议进行了综合和扩展，提出了一个具有分布式特点的大型动态多播群组密钥管理方案，并给出了它的算法．对密钥服务器的存储量、加密计算量及通信量等的分析结果表明，该方案具有良好的有效性和可扩展性，适用于大型的多播群组．     

网络时间协议实现分布式系统内时钟同步的原理分析

在某些关键应用中，分布式系统对系统内时钟的一致性要求是比较高的．网络时间协议作为一个Internet标准协议，可以作为分布式系统时钟同步的有效工具．本文介绍了网络时间协议的基本模型和体系结构，并着重分析了使用网络时间协议实现时钟同步的基本原理．     

分布存储媒体传输的同步算法

分布多媒体数据库（ＤＭＤＢ）和分布多媒体信息系统（ＤＭＩＳ）的一个重要需求是多媒体同步传输．文中先分析了在ＤＭＤＢ和ＤＩＭＳ中多媒体同步传输的特点和要求，然后提出了一个能满足这些要求的同步传输方案．在这个方案中，多媒体同步关系是用动态同步Ｐｅｔｒｉ网（ＤＳＰＮ）来显式描述的；通过文中给出的同步传输算式和同步发送算法，可以在ＤＳＰＮ模型的基础上产生同步传输调度方案；同时接收端能根据实际通信状况和多媒体的同步要求，对来自多个服务器的多媒体对象进行动态同步控制．     

Protocol considerations for a prefix-caching proxy for multimedia streams

The increasing popularity of multimedia streaming applications introduces new challenges in content distribution. Web-initiated multimedia streams typically experience high start-up delay, due to large protocol overheads and the poor delay, throughput, and loss properties of the Internet. Internet service providers can improve performance by caching the initial segment (the prefix) of popular streams at proxies near the requesting clients. The proxy can initiate transmission to the client while simultaneously requesting the remainder of the stream from the server. This paper analyzes the challenges of realizing a prefix-caching service in the context of the IETF's Real-Time Streaming Protocol (RTSP), a multimedia streaming protocol that derives from HTTP. We describe how to exploit existing RTSP features, such as the Range header, and how to avoid several round-trip delays by caching protocol information at the proxy. Based on our experiences, we propose extensions to RTSP that would ease the development of new multimedia proxy services. In addition, we discuss how caching the partial contents of multimedia streams introduces new challenges in cache coherency and feedback control. Then, we briefly present our preliminary implementation of prefix caching on a Linux-based PC, and describe how the proxy interoperates with the RealNetworks server and client.     

PLUS: a probe-loss utilization streaming mechanism for distributed multimedia presentation systems

We present a new flow and congestion control scheme, PLUS (Probe-Loss Utilization Streaming protocol), for distributed multimedia presentation systems. This scheme utilizes probing of the network situation and an effective adjustment mechanism to data loss to support multimedia presentations. The proposed scheme is also designed to scale with increasing number of PLUS-based streaming traffic and to live in harmony with TCP-based traffic. The novelty of the PLUS protocol is that it utilizes the knowledge of its future bottleneck bandwidth in probing the current network situation. This can be achieved by a priori knowledge of the multimedia data before a presentation is requested by a client. Compression schemes like MPEG introduce dependencies on media units. I frames are needed to successfully decode P and B frames, and P frames are needed to decode B frames. A loss of an I or P frame automatically eliminates dependent media units. Our probing scheme increases the successful transmission of critical I and P packets without the overhead of error-correction-schemes. Probing is done using B-frame packets. The advantage is that we use data packets as probe packets. With the PLUS protocol we address the need to avoid congestion rather than react to it. Experiments demonstrate the effectiveness of the approach in utilizing network resources and decreasing loss ratios.     

Simulation and Analysis of Packet Loss in User Datagram Protocol Transfers

Understanding packet loss patterns in Internet Protocol (IP) networks is important for achieving the desired quality of service in multimedia transfers. In this paper, we study the loss patterns in video transfers using User Datagram Protocol (UDP) in a congested packet network. We use trace-driven ns-2 simulations to collect packet loss traces in networks, and we apply wavelet analysis to investigate the behavior of packet loss on various time-scales. We show that time-scales are essential for understanding loss behavior and that packet loss exhibits long-range dependence over the coarser time-scales.     

A model for specification and synchronization of data for distributed multimedia applications

As network technology provides the capability to handle multimedia traffic and the demand of multimedia services increases, protocols are required for effective communication of multimedia data in a distributed environment. Synchronization is one of the key issues in a multimedia system. Most of the current approaches do not support an integrated solution to the problem of synchronization. In this paper we propose a mechanism for synchronization of multimedia data in distributed environment where the accuracy of the protocol can be tailored to the application. The system model supports live and video-on-demand service. We present a scheme where the specification of the temporal requirements provided by the application can be directly mapped to obtain the information necessary to enforce the synchronization required. We present two examples of specifying the temporal requirements and process of obtaining the information and present performance results of our simulation studies.     

基于RTP/RTCP协议的实时数据传输与同步控制策略

针对分组交换网络中的实时媒体传输,考虑非QOS保证的分组网络可能带来的传输丢包、乱序和抖动等情况,采用基于RTP/RTCP协议的媒体传输和媒体控制机制,在媒体流中添加时间戳等控制信息,通过播放时延控制算法进行媒体内同步,并在媒体内同步的基础上,根据发送方的绝对时间戳和RTP时间戳的对应关系,确定不同媒体流之间的同步点,从而达到多通道媒体间同步的效果。     

一种适合窄带网络多媒体教学的文件传输协议──NFTP的研究与实现

目前在网络环境下进行多媒体教学,网络速度是一个严重的瓶颈问题。在分析瓶颈产生的基础上,作者提出了一个新的文件传输协议──ＮＦＴＰ（Ｎａｒｒｏｗｂａｎｄ　Ｆｉｌｅ　Ｔｒａｎｓｆｅｒ　Ｐｒｏｔｏｃｏｌ窄带文件传输协议）,利用广播数据包方法解决多媒体教学中的速度问题,并设计了一组协议原语,用于协议的描述。     

基于SIP协议的WebRTC信令研究与应用

在近年来随着用户对音视频通话质量要求的提高,WebRTC以其强大的多媒体处理能力得到了广泛的应用.然而WebRTC提供的JSEP是一种弱信令,在企业级的融合通信应用中必须将WebRTC与实际的信令协议相结合.SIP是IMS的核心技术,对多媒体会话的控制起着非常重要的作用.本文介绍了WebRTC和SIP协议融合的已有方案,研究了WebRTC和SIP协议互通需要解决的问题,提出了一种WebRTC的PeerConnection层和SIP协议在客户端的融合方案,并和其他方案对比,得出该方案的优缺点.     

Energy efficient and reliable data delivery in urban sensing applications: A performance analysis

Urban sensing is an emerging application field for Wireless Sensor Networks (WSNs), where a number of static sensors is sparsely deployed in an urban area to collect environmental information. Data sensed by each sensor are, then, opportunistically transmitted to Mobile Nodes (MNs) that happen to be in contact. In the considered scenario, communications between MNs and sensors require paradigms with a minimal synchronization between devices, extremely fast and energy efficient, especially at the sensor side. To deal with the above issues, in [1] we proposed a hybrid protocol for data delivery from sensors to MNs, named Hybrid Adaptive Interleaved Data Protocol (HI). By combining Erasure Coding (EC) with an Automatic Repeat reQuest (ARQ) scheme, the proposed protocol maximizes the reliability of communications while minimizing the energy consumed by sensors. In this paper, we present an in-depth analysis of the HI performance. We provide an analytical evaluation by defining a flexible model to derive the probability of data delivery and exploiting it to investigate the performance over a wide range of parameters. Moreover, we perform an experimental study to evaluate the HI effectiveness on real sensor platforms. Specifically, we analyze the impact of resource constraints imposed by sensors on data delivery and provide a careful characterization of its actual consumption of resources.     

A Service-Centric Multicast Architecture and Routing Protocol

In this paper, we present a new multicast architecture and the corresponding multicast routing protocol for providing efficient and flexible multicast services over the Internet. Traditional multicast protocols construct and update the multicast tree in a distributed manner, which may cause two problems: first, since each node has only local or partial information on the network topology and group membership, it is difficult to build an efficient multicast tree and, second, due to the lack of complete information, broadcast is often used for sending control packets and data packets, which consumes a great deal of network bandwidth. In the newly proposed multicast architecture, a few powerful routers, called m-routers, collect multicast-related information and process multicast requests based on the information collected. The m-routers handle most of the multicast-related tasks, whereas other routers in the network only need to perform minimum functions for routing. The m-routers are designed to be able to handle simultaneous many-to-many communications efficiently. The new multicast routing protocol, called the Service-Centric Multicast Protocol (SCMP), builds a shared multicast tree rooted at the m-router for each group. The multicast tree is computed in the m-router by employing the Delay-Constrained Dynamic Multicast (DCDM) algorithm, which dynamically builds a delay-constrained multicast tree and minimizes the tree cost as well. The physical construction of the multicast tree over the Internet is performed by a special type of self-routing packets in order to minimize the protocol overhead. Our simulation results on ns-2 demonstrate that the new SCMP protocol outperforms other existing protocols and is a promising alternative for providing efficient and flexible multicast services over the Internet.