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A Call Control and Multi-party usage Framework for the Session Initiation Protocol (SIP), (2003)

by Rohan Mahy
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The Session Initiation Protocol (SIP) "Join" Header", RFC 3911

by R. Mahy, D. Petrie , 2004
"... Status of this Memo ..."
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...l Copyright Statement . . . . . . . . . . . . . . . . . . 17 1. Introduction This document describes a SIP [1] extension header field as part of the SIP multiparty applications architecture framework =-=[12]-=-. The Join header is used to logically join an existing SIP dialog with a new SIP dialog. This is especially useful in peer-to-peer call control environments. One use of the "Join" header is to insert...

SIP Telephony Device Requirements and Configuration", RFC 4504

by H. Sinnreich, C. Stredicke , 2006
"... This memo provides information for the Internet community. It does not specify an Internet standard of any kind. Distribution of this memo is unlimited. Copyright Notice Copyright (C) The Internet Society (2006). This document describes the requirements for SIP telephony devices, based on the deploy ..."
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This memo provides information for the Internet community. It does not specify an Internet standard of any kind. Distribution of this memo is unlimited. Copyright Notice Copyright (C) The Internet Society (2006). This document describes the requirements for SIP telephony devices, based on the deployment experience of large numbers of SIP phones and PC clients using different implementations in various networks. The objectives of the requirements are a well-defined set of interoperability and multi-vendor-supported core features, so as to enable similar ease of purchase, installation, and operation as found for PCs, PDAs, analog feature phones or mobile phones. We present a glossary of the most common settings and some of the more widely used values for some settings.
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... devices MUST support the Third Party Call Control model [19], in the sense that they may be the controlled device. Req-35: SIP telephony devices SHOULD support SIP call control and multi-party usage =-=[20]-=-. Req-36: SIP telephony devices SHOULD support conferencing services for voice [21], [22] and interactive text [23] and if equipped with an adequate display MAY also support instant messaging (IM) and...

Generalized third-party call control in SIP networks

by Eric Cheung, Pamela Zave - In Proceedings of the Second International Conference on Principles, Systems and Applications of IP Telecommunications. Springer-Verlag LNCS , 2008
"... Abstract. Third-party call control (3PCC) is essential to implementing advanced services in Voice-over-IP (VoIP) networks. It allows intermediary applications to control how the media streams of endpoint devices are connected together. However the Session Initiation Protocol (SIP), the widely adopte ..."
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Abstract. Third-party call control (3PCC) is essential to implementing advanced services in Voice-over-IP (VoIP) networks. It allows intermediary applications to control how the media streams of endpoint devices are connected together. However the Session Initiation Protocol (SIP), the widely adopted open standard for VoIP signaling, presents a number of challenges that make 3PCC in SIP complex and prone to errors. Previously proposed solutions only address operations under ideal conditions and thus are incomplete. Furthermore, the problem of compositional 3PCC by multiple applications have not been addressed. In this paper, we propose a general solution for robust and comprehensive media connectivity control. The solution has been verified, and allows multiple applications operating concurrently in a call path to interoperate successfully. 1
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...control in SIP has received a significant amount of attention and is recognized as an important call control model. In particular the clickto-dial use case has been cited extensively (e.g. [10], [6], =-=[9]-=-). However, thus far the literature has only included limited example call flows under ideal conditions. Moreover, the click-to-dial use case is relatively simple as the endpointsinvolved are restric...

Peer-to-Peer Videoconferencing with H.264 Software Codec for Mobiles

by Hans L. Cycon, Thomas C. Schmidt, Gabriel Hege, Matthias Wählisch, Detlev Marpe, Mark Palkow
"... A rapidly growing number of carriers offer wireless video services to their customers, taking advantage of high quality video codecs implemented in dedicated hardware of selected mobile devices. In this paper we introduce a video conferencing software, which seamlessly integrates mobile with station ..."
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A rapidly growing number of carriers offer wireless video services to their customers, taking advantage of high quality video codecs implemented in dedicated hardware of selected mobile devices. In this paper we introduce a video conferencing software, which seamlessly integrates mobile with stationary users in a provider and device independent fashion. Innovations of this work are twofold. At first we report on a mobile realization of an H.264 video codec and its performance on a standard consumer Smartphone. Operating within the tight bounds of real-time compliance on mobiles, this software is an adapted version of a highly optimized H.264 codec. This DAVC codec, which we introduce along the line, significantly outperforms compatible H.264 realizations and allows for a scalable adaptation of its frame rate. In the second part we present a barrier-resistant peer-to-peer group communication scheme, which scales well for medium-size conferences and accounts for the heterogeneous nature of mobile and stationary participants. An outlook on mobility related group communication issues and future optimizations based on structured communication layers concludes the work. Index Terms Mobile video coding, mobile conferencing, peer-topeer group communication, distributed SIP conference management 1.
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... assistance. Such a solution requires group session management and media distribution at peers, which for the sake of standard compliance we realize with group conferencing functions in SIP, cf. [8], =-=[9]-=-, [10]. Implemented as pure software on standard personal devices, user agent peers are exposed to severe restrictions inFigure 3. The mobile video application. real-world deployments: Often they are...

Session Initiation Protocol (SIP) Call Control- Transfer

by R. Sparks, A. Johnston, D. Petrie, Sipez Llc , 2009
"... This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. Distribution of this memo is unlimited. Copyright Notice Copyright (c) 2009 IETF Trust and the persons identified as the ..."
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This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. Distribution of this memo is unlimited. Copyright Notice Copyright (c) 2009 IETF Trust and the persons identified as the
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...C 5589sSIP CC TransfersJune 2009s1.sOverviewsThis document describes providing Call Transfer capabilities andsrequirements in SIP [RFC3261].sThis work is part of the multipartyscall control framework =-=[CC-FRMWRK]-=-.sThe mechanisms discussed here are most closely related tostraditional, basic, and consultation hold transfers.sThis document details the use of the REFER method [RFC3515] andsReplaces [RFC3891] head...

The Authentication and Processing Performance of Session Initiation Protocol (SIP) Based Multi-party Secure Closed Conference System

by Jongkyung Kim , Hyuncheol Kim , Seongjin Ahn , Jinwook Chung , 2004
"... Abstract. This paper presents, a new SIP based multi-party secure closed conference system. In the traditional participants, except of captain, conference system doesn't have a privilege for UA (User Agent) that accepts or declines new participants. On the other hand, closed conference system ..."
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Abstract. This paper presents, a new SIP based multi-party secure closed conference system. In the traditional participants, except of captain, conference system doesn't have a privilege for UA (User Agent) that accepts or declines new participants. On the other hand, closed conference system supports this function. We also propose for closed conference system authentication procedure. By means of a real implementation, we provide an experimental performance analysis of SIP security mechanisms.
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...P message contains information that a user or server wishes to keep private. Securing SIP header and body information can be motivated by two reasons. One is that we need maintain private user and network information in order to guarantee a certain level of privacy. Another is that we have to avoid SIP sessions being set up or changed by attackers faking the identity of someone else. The SIP UAC (User Agent Client), calling side, can identity itself to a UAS (User Agent Server), called side. Therefore, SIP authentication applies only to user-to-user, user-proxy or user-registrar communication [3] A representation of the SIP digest authentication procedure is given in Fig. 2. The function F() used to compute the response specifies how to combine the input paJ. Kim et al. 726 rameter with some iterations of a digest algorithm [4][5]. The authentication procedure is run when the UAS, a proxy server, or the registrar server requires the calling side (UAC) to be authenticated before accepting the call, forwarding the call, or accepting the registration. MCP(Multipoint Control Unit) Conference Bridge + Media Bridge Signalling Media SIP UA SIP UA SIP UA SIP UA Signalling Media SIP UA SIP UA ...

The Session Inititation Protocol (SIP) "Replaces " Header

by R. Mahy, B. Biggs, R. Dean , 2003
"... draft-ietf-sip-replaces-04.txt ..."
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draft-ietf-sip-replaces-04.txt

Multiservice Switching Forum Contribution Contribution Number: msf2004.173.00 Last Saved: (10 h August 2004) Working Group: Interoperability and Test Title: Test Plan for Scenario 2. Source: MSF MSF Confidential

by unknown authors
"... Scenario 2 test plan msf2004.173.00 ..."
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Scenario 2 test plan msf2004.173.00
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...Session Initiation Protocol,” IETF RFC3261, Jun. 2002. [7] J. Rosenberg, “Best Current Practice for Third Party Cal Control in the Session Initiation Protocol,” draft-ietf-sipping-3pcc-06, Dec, 2003. =-=[8]-=- Rohan Mahy, "A Call Control and Multi-party usage Framework for the Session Initiation Protocol (SIP),” Sep.2003. [9] Eric VanWagner and Mike Huang, “Implementation Agreement for MEGACO/H.248 Profile...

Call Control- Conferencing for User Agents

by A. Johnston, O. Levin , 2006
"... This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. Distribution of this memo is unlimited. Copyright Notice Copyright (C) The Internet Society (2006). This specification defines conferencing call control fe ..."
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This document specifies an Internet Best Current Practices for the Internet Community, and requests discussion and suggestions for improvements. Distribution of this memo is unlimited. Copyright Notice Copyright (C) The Internet Society (2006). This specification defines conferencing call control features for the Session Initiation Protocol (SIP). This document builds on the Conferencing Requirements and Framework documents to define how a tightly coupled SIP conference works. The approach is explored from the perspective of different user agent (UA) types: conferenceunaware, conference-aware, and focus UAs. The use of Uniform Resource Identifiers (URIs) in conferencing, OPTIONS for capabilities discovery, and call control using REFER are covered in detail with example call flow diagrams. The usage of the isfocus feature tag is
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...entially have joined the conference by different means (i.e., dial-in, dial-out, scheduled, or ad-hoc). The call control and dialog manipulation approach is based on the multiparty framework document =-=[15]-=-. That document defines the basic approach of service design adopted for SIP, which includes the following: - Definition of primitives, not services - Signaling model independent - Invoker oriented - ...

Adobe

by H. Sinnreich, A. Johnston, E. Shim, K. Singh , 2009
"... Simple SIP Usage Scenario for Applications in the Endpoints For Internet-centric usage, the number of SIP-required standards for presence and IM and audio/video communications can be drastically smaller than what has been published by using only the rendezvous and session-initiation capabilities of ..."
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Simple SIP Usage Scenario for Applications in the Endpoints For Internet-centric usage, the number of SIP-required standards for presence and IM and audio/video communications can be drastically smaller than what has been published by using only the rendezvous and session-initiation capabilities of SIP. The simplification is achieved by avoiding the emulation of telephony and its model of the intelligent network. ’Simple SIP ’ relies on powerful computing endpoints. Simple SIP desktop applications can be combined with rich Internet applications (RIAs). Significant telephony features may also be implemented in the endpoints. This approach for SIP reduces the number of SIP standards with which to comply-- from roughly 100 currently, and still growing, to about
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...endpoints can, however, accomplish most telephony functions as well. This has been amply documented in SIPrelated work in the IETF, such as: o "A Call Control and Multi-party usage framework for SIP" =-=[19]-=- presents a large assortment of telephony applications where the call control resides in the participating endpoints that use the peer-to-peer feature invocation model. The peer-to-peer design and its...

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