Definition of Events for Channel-Oriented Telephony Signalling (original) (raw)

Definition of Events for Modem, Fax, and Text Telephony Signals

2006

Status of This Memo This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards" (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited.

RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals Status of this Memo

2000

Status of this Memo This document specifies an Internet standards track protocol for the Internet community, and requests discussion and suggestions for improvements. Please refer to the current edition of the "Internet Official Protocol Standards" (STD 1) for the standardization state and status of this protocol. Distribution of this memo is unlimited.

Signaling for Internet telephony

Proceedings Sixth International Conference on Network Protocols (Cat. No.98TB100256)

Internet telephony must offer the standard telephony services. However, the transition to Internetbased telephony services also provides an opportunity to create new services more rapidly and with lower complexity than in the existing public switched telephone network (PSTN). The Session Initiation Protocol (SIP) is a signaling protocol that creates, modifies and terminates associations between Internet end systems, including conferences and point-to-point calls. SIP supports unicast, mesh and multicast conferences, as well as combinations of these modes. SIP implements services such as call forwarding and transfer, placing calls on hold, camp-on and call queueing by a small set of call handling primitives. SIP implementations can re-use parts of other Internet service protocols such as HTTP and the Real-Time Stream Protocol (RTSP). In this paper, we describe SIP, and show how its basic primitives can be used to construct a wide range of telephony services. may not be generating media streams. For example, when a lecture is multicast, the initiator of the call may not be sending or receiving data. We see Internet telephony signaling encompassing a number of functions: Name translation and user location involves the mapping between names of different levels of abstraction, e.g., a common name at a domain and a user name at a particular Internet host. These translations may involve simple table lookups at the server or may involve locating the party, as described in Section 4.2. Feature negotiation allows a group of end systems to agree on what media to exchange and their respective parameters such as encodings. The set and type of media need not be uniform within a call, as different point-to-point sessions may involve different media and media parameters. Many software codecs are able to receive different encodings within a single conference and in parallel, for example, while being restricted to sending one type of media for each stream. Any call participant can invite others into an existing call and terminate associations with some (call participant management). During the call, participants should be able to transfer and hold other users. The most general model of a multi-party association is that of a full or partial mesh of invitations, with the possible addition of multicast distribution between some or all participants. Feature changes make it possible to adjust the composition of media sessions during the course of a call, either because the participants require additional or reduced functionality or because of constraints imposed or removed by the addition or removal of call participants. Not all of these functions have to be addressed by one protocol. For example, H.323 may be used to establish sessions between the end system and the gateway, while the Session Initiation Protocol (SIP), the protocol described here, might be responsible for gateway-to-gateway signaling. Other conference management functions are beyond the scope of signaling. These include distributed queue management for floor control and distributed counting for voting. Unlike signaling, both of these require some form of reliable multicast; however, for small groups, a multipoint controller can perform the replication. Signaling can be used to introduce this functionality into conferences as needed. In Section 2, we discuss the basic architecture for telephony signaling services. In Section 3 we discuss the basics of SIP operation, its addressing structure, message syntax and transport. In Section 4, we discuss how SIP can be used for telephony services, focusing on how services are constructed from simple primitive tools. We then briefly discuss, in Section 5, the interaction of telephony signaling with stored media control protocols. We then mention related work in Section 6, and conclude in Section 7. 2 Internet Telephony Architecture 2.1 Separation of Signaling Functionality Unlike circuit-switched telephony, Internet telephony services are built on a range of packet switched protocols, as illustrated in Fig. 1. For example, the functionality of the SS7 telephony signaling protocol encompasses routing, resource reservation, call admission, address translation, call establishment, call management and billing. In an Internet environment, routing is handled by protocols such as BGP [2], resource reservation by RSVP [3] or other resource reservation protocols [4]. SIP, described here, translates application-layer addresses, establishes and manages calls. There is currently no Internet telephony billing protocol in the Internet, although RADIUS [5], in combination with SIP authentication, may initially serve that purpose. This separation of concerns affords greater architectural flexibility. For example, Internet telephony may be used without per-call resource reservation in networks with sufficient capacity; billing may not be necessary in a PBX-like environment. On the other hand, removing the "atomicity" of call setup found in the current telephone system also breaks assumptions: since call setup and resource reservation are distinct, one may succeed, while the other may fail. If resources are reserved first, the caller may incur a cost for

Problem verification during execution of H.323 signaling

Like every other network protocols and technologies, VoIP implementation must be kept within certain limits that communication services are on expected and acceptable level. For H.323 VoIP networks, there are two areas of performance which interest network engineers. Signaling performance affects length of time necessary for call establishment between two H.323 terminals. RTP performance affects delay and overall clarity of voice signal during VoIP conversation. This paper gives an overview of RTP performance and shows how to use H.323 decoded data in order to evaluate signaling performance.

A Session Initiation Protocol (SIP) Event Package for Communication Diversion Information

2008

This draft defines a Session Initiation Protocol (SIP) Event Notification Framework-based mechanism for notifying Users about diversions (re-directions or forwarding) of their incoming communication sessions. A new event package is proposed for allowing users to subscribe for and receive such notifications. Users have further capability to define filters controlling the selection, rate and content of such notifications. The applicability of this event package includes 3GPP IMS.

EVA: an event-based framework for developing specialised communication protocols

Proceedings IEEE International Symposium on Network Computing and Applications. NCA 2001, 2001

This paper presents a framework for the development of higher level communication protocols that provides extra functionalities (e.g. one-to-many ordered delivery, atomic delivery, etc.) not supplied by standard off-the-shelf lower level communication protocols (e.g. the TCP/IP suite of protocols). The framework is based on the event channel abstraction which allows circumventing the main drawbacks of the layered-based approach traditionally used to develop such protocols, whilst at the same time providing a flexible, simple and well structured way to implement them. The event channel service provided by EVA establishes how entities that share the same address space interact. Then, the application designer has the opportunity to define the most appropriate lower level communication protocols that control the way entities that execute within different processes will interact. The framework specifies a way to accommodate these protocols and provides several standard protocol implementations. Further, it is described a development methodology for constructing applications (specialised communication protocols) on top of the framework. In designing the framework, we have followed the approach of using, whenever possible, well established concepts (e.g. event notification service, design patterns, etc.), thus the paper also discusses the utilisation of such concepts in improving both the efficiency and the structuring of the framework and of the applications to be built on top of it.

Access signalling — Q.2931 and related standards

Bt Technology Journal, 1998

This paper provides a brief description of the International Telecommunications Union (ITU-T) specified access signalling protocol Q.2931 and the supporting layer 2 signalling ATM adaptation layer (SAAL) that provides the release 1 signalling mechanism Digital Signalling System No 2 (DSS2). In addition to identifying the functions that are required to be incorporated in a signalling protocol to support call/connection establishment, in-call control and clear-down for a basic point-to-point call, the extensions required for a point-to-multipoint call are also considered. The paper then considers further extensions that have been produced to enable the negotiation, modification and look-ahead in relation to the connection in order to provide the capability set 2 step 1 DSS2 signalling mechanism. In conclusion, a simple road map to the DSS2 ITU-T standards in provided.

Telecommunications Protocols Fundamentals

Telecommunication Systems [Working Title], 2019

The need for communication amongst people and electrical systems motivated the emergence of a large number of telecommunications protocols. The advances in digital networks and the internet have contributed to the evolution of telecommunications worldwide. The purpose of this chapter is to provide students and researchers with a clear presentation of telecommunications core protocols that are utilised in different research domains including telephony, brain-computer interface (BCI) and voice and digital telecommunications. Indeed, BCI involves different electrical signals, communications concepts and telecommunications protocols. This chapter introduces the reader to the core concepts in communications including analogue and digital telecommunications protocols that are utilised generally in communications and in particular in BCI systems. The topics covered in this chapter include telecommunications protocols, communications media, electrical signals, analogue and digital modulation techniques in digital communications, softwaredefined radio, overview on 10-Mbps Ethernet protocol and Session Initiation Protocol (SIP).