Handbook on Session Initiation Protocol

Networked Multimedia Communications for IP Telephony

Radhika Ranjan Roy

• Publisher: CRC Press
• Year: 2018
• eBook ISBN: 9781315359922
• Edition: 1
• Topic: Computer Science
• Subtopic: Information Technology
• Index: Computer Science

Session Initiation Protocol (SIP), standardized by the Internet Engineering Task Force (IETF), has emulated the simplicity of the protocol architecture of hypertext transfer protocol (HTTP) and is being popularized for VoIP over the Internet because of the ease with which it can be meshed with web services. However, it is difficult to know exactly how many requests for comments (RFCs) have been published over the last two decades in regards to SIP or how those RFCs are interrelated.

Handbook on Session Initiation Protocol: Networked Multimedia Communications for IP Telephony solves that problem. It is the first book to put together all SIP-related RFCs, with their mandatory and optional texts, in a chronological and systematic way so that it can be used as a single super-SIP RFC with an almost one-to-one integrity from beginning to end, allowing you to see the big picture of SIP for the basic SIP functionalities. It is a book that network designers, software developers, product manufacturers, implementers, interoperability testers, professionals, professors, and researchers will find to be very useful.

The text of each RFC from the IETF has been reviewed by all members of a given working group made up of world-renowned experts, and a rough consensus made on which parts of the drafts need to be mandatory and optional, including whether an RFC needs to be Standards Track, Informational, or Experimental. Texts, ABNF syntaxes, figures, tables, and references are included in their original form. All RFCs, along with their authors, are provided as references. The book is organized into twenty chapters based on the major functionalities, features, and capabilities of SIP.

Chapter 1
Networked Multimedia Services

Abstract

Networked multimedia services have some key technical characteristics that need to be satisfied by any multimedia signaling and media protocols that deal with multimedia services over the networks. The characteristics of real-time, near-real-time, and non-real-time services, especially focusing on their performances, are described.

1.1 Introduction

Multimedia applications consisting of audio, video, and/or data that provide communications services at a distance, connected over networks, are termed networked multimedia services. These applications can be distributed across networks, and the communication connectivity can be one-to-one, many-to-many, many-to-one, and one-to-many over the networks. The networking function embedded into multimedia applications makes them networked multimedia services. The network architecture that facilitates networked multimedia services can be termed the networked multimedia services architecture.

1.2 Functional Characteristics

The functional characteristics of networked multimedia applications can be very complex. Multimedia communications for conversation applications will not only need to be in real-time, meeting the stringent performance requirements, but also a simple point-to-point audio call may evolve, at users’ discretion, to become a multimedia call, or a multipoint multimedia call if more participants are added into the call [1,2]. If the data-sharing application is added into the same call, the situation becomes more complicated as it is expected to be the usual case for real-time multimedia collaboration. The connectivity of the communications between the participants can vary from point-to-point to many-to-many, usually with symmetric traffic flows. In the case of video-on-demand (VOD) applications, the connectivity will primarily be from one-to-many with highly asymmetric traffic flows because the video applications are usually distributed from the centralized multimedia servers to multiple users, although the performance constraints may be a little less stringent from those of the conversational applications. The functional characteristics of some multimedia applications, such as multimedia messaging, can be both one-to-many and many-to-many with respect to connectivity, while the traffic flows can also be symmetric and asymmetric depending on the needs of the participants. In general, the functional characteristics of all other multimedia applications may typically fall within these three categories explained here.

1.3 Performance Characteristics

The audio and video of multimedia applications can be continuous, while data consisting of text, still images, and/or graphics can be discrete. However, animation that is also considered as a part of data is continuous, consisting of audio, video, still images, graphics, and/or texts, and needs inter-/intramedia synchronization. Audio, video, and still images are usually captured from the real world, while text, graphics, and animation are synthesized by computers.

On the basis of the performance characteristics of communications, multimedia applications can be categorized as follows: real-time (RT), near-real-time (near-RT), and non-real-time (non-RT). RT applications will have strict bounds on packet loss, packet delay, and delay jitter, while near-RT applications will have less strict bounds on those performance parameters than those of the RT applications. For example, teleconferencing (TC) and video teleconferencing (VTC)/videoconferencing (VC) are considered RT services because of real-time two-way, point-to-point/multipoint conversations between users and, the audio and video performance requirements can be stated as follows [1]:

■ One-way end-to-end delay (including propagation, network, and equipment) for audio or video should be between 100 and 150 ms.

■ Mean-opinion-score (MOS) level for audio should be between 4.0 and 5.0.

■ MOS level for video should be between 3.5 and 5.0.

■ End-to-end delay jitter should be very short, less than 250 μs in some cases.

■ Bit error rate (BER) should be very low for good quality audio or video, although some BER can be tolerated.

■ Intermedia and intramedia synchronization need to be maintained using suitable algorithms.

■ Differential delay between audio and video transmission should be between no more than −20 ms to +40 ms for maintaining proper intermedia synchronization.

One-way VOD [2], which is considered a near-RT communication, can have much less stringent performances than those of TC or VTC. The text or graphics are non-RT applications, and the one-way delay requirement can be of the order of a few seconds; however, unlike audio or video, it cannot tolerate any BER.

The synchronization requirements between different media of multimedia applications impose a heavy burden on the multimedia transport networks, especially for the packet networks such as the Internet Protocol (IP). RT applications are also considered live multimedia applications with the generation of live audio, video, and/or data from live sources of microphones, video cameras, and/or application sharing by human/machine, while near-RT applications are usually retrieved from databases and can be considered as retrieval multimedia applications. Consequently, the synchronization requirements between RT and near-RT applications are also significantly different. The transmission side of the RT applications does not require much control, while near-RT applications must have some defined relationships between media and require some scheduling mechanisms for guaranteed synchronizat...

Table of contents

• Cover
• Half Title
• Title Page
• Copyright Page
• Dedication
• Table of Contents
• List of Figures
• List of Tables
• Preface
• Author
• 1 Networked Multimedia Services
• 2 Basic Session Initiation Protocol
• 3 SIP Message Elements
• 4 Addressing in SIP
• 5 SIP Event Framework and Packages
• 6 Presence and Instant Messaging in SIP
• 7 Media Transport Protocol and Media Negotiation
• 8 DNS and ENUM in SIP
• 9 Routing in SIP
• 10 User and Network-Asserted Identity in SIP
• 11 Early Media in SIP
• 12 Service and Served-User Identity in SIP
• 13 Connections Management and Overload Control in SIP
• 14 Interworking Services in SIP
• 15 Resource Priority and Quality of Service in SIP
• 16 Call Services in SIP
• 17 Media Server Interfaces in SIP
• 18 Multiparty Conferencing in SIP
• 19 Security Mechanisms in SIP
• 20 Privacy and Anonymity in SIP
• Appendix A: ABNF
• Appendix B: Reference RFCs
• Index