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Marzo). Streaming video over the Internet: approaches and directions.
- IEEE Transactions on Circuits and Systems for Video Technology,
, 2001
"... ..."
Real time streaming protocol (RTSP
, 1998
"... This document is an Internet-Draft. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet-Drafts. Internet-Drafts are draft documents valid for a maximum of six ..."
Abstract
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Cited by 118 (10 self)
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This document is an Internet-Draft. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet-Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as “work in progress”. To learn the current status of any Internet-Draft, please check the “1id-abstracts.txt ” listing contained in the Internet-Drafts Shadow Directories on ftp.is.co.za (Africa), nic.nordu.net (Europe), munnari.oz.au (Pacific Rim), ds.internic.net (US East Coast), or ftp.isi.edu (US West Coast). Distribution of this document is unlimited. Copyright Notice Copyright (c) The Internet Society (2003). All Rights Reserved. This memorandum is a revision of RFC 2326, which is currently a Proposed Standard. The Real Time Streaming Protocol, or RTSP, is an application-level protocol for control over the
YESSIR: A simple reservation mechanism for the Internet
- In International Workshop on Netwwork and Operating Systems Support for Digital Audio and Video (NOSSDAV97
, 1998
"... RSVP has been designed to support resource reservation in the Internet. However, it has two major problems: complexity and scalability. The former results in large message processing overhead at end systems and routers, and inefficient firewall processing at the edge of the network. The latter impli ..."
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Cited by 101 (22 self)
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RSVP has been designed to support resource reservation in the Internet. However, it has two major problems: complexity and scalability. The former results in large message processing overhead at end systems and routers, and inefficient firewall processing at the edge of the network. The latter implies that in a backbone environment, the amount of bandwidth consumed by refresh messages and the storage space that is needed to support a large number of flows at a router are too large. We have developed a new reservation mechanism that simplifies the process of establishing reserved flows while preserving many unique features introduced by RSVP. Simplicity is measured in terms of control message processing, data packet processing, and user-level flexibility. Features such as robustness, advertising network service availability and resource sharing among multiple senders are also supported in the proposal. The proposed mechanism, YESSIR (YEt another Sender Session Internet Reservations) generates reservation requests by senders to reduce the processing overhead, builds on top of RTCP, uses soft state to maintain reservation states, supports shared reservation and associated flow merging and is compatible with the IETF Integrated Services models. YESSIR extends the all-or-nothing reservation model to support partial reservations that improve over the duration of the session. To address the scalability issue, we investigate the possibility of using YESSIR for per-stream reservation and RSVP for aggregate reservation.
A Comparison of SIP and H.323 for Internet Telephony
"... Two standards have recently emerged for signaling and control for Internet Telephony. One is ITU Recommendation H.323, and the other is the IETF Session Initiation Protocol (SIP). These two protocols represent very different approaches to the same problem: H.323 embraces the more traditional circuit ..."
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Cited by 74 (8 self)
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Two standards have recently emerged for signaling and control for Internet Telephony. One is ITU Recommendation H.323, and the other is the IETF Session Initiation Protocol (SIP). These two protocols represent very different approaches to the same problem: H.323 embraces the more traditional circuit-switched approach to signaling based on the ISDN Q.931 protocol and earlier H-series recommendations, and SIP favors the more lightweight Internet approach based on HTTP. In this paper, we compare SIP and H.323 on complexity, extensibility, scalability, and features.
Internet telephony: Architecture and protocols – an IETF perspective
- Computer Networks and ISDN Systems
, 1999
"... 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 ..."
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Cited by 71 (20 self)
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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. 1
Programming internet telephony services
- IEEE Network
, 1999
"... Internet telephony enables a wealth of new service possibilities. Traditional telephony services, such as call forwarding, transfer, and 800 number services can be enhanced by interaction with email, web, and directory services. Additional media types, like video and interactive chat, can be added a ..."
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Cited by 58 (27 self)
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Internet telephony enables a wealth of new service possibilities. Traditional telephony services, such as call forwarding, transfer, and 800 number services can be enhanced by interaction with email, web, and directory services. Additional media types, like video and interactive chat, can be added as well. One of the challenges in providing these services is how to effectively program them. Programming these services requires decisions regarding where the code executes, how it interfaces with the protocols that deliver the services, and what level of control the code has. In this paper, we consider this problem in detail. We develop requirements for programming Internet telephony services, and we show that at least two solutions are required — one geared for service creation by trusted users (such as administrators), and one geared for service creation by untrusted users (such as consumers). We review existing techniques for service programmability in the Internet and in the telephone network, and extract the best components of both. The result is a Common Gateway Interface (CGI) that allows trusted users to develop services, and the Call Processing Language (CPL) that allows untrusted users to develop services. 1
A Comparison of Hard-state and Soft-state Signaling Protocols
- In: Proc. of SIGCOMM 2003
, 2003
"... One of the key infrastructure components in all telecommunication networks, ranging from the telephone network, to VC-oriented data networks, to the Internet, is its signaling system. Two broad approaches towards signaling can be identified: so-called hard-state and soft-state approaches. Despite th ..."
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Cited by 45 (1 self)
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One of the key infrastructure components in all telecommunication networks, ranging from the telephone network, to VC-oriented data networks, to the Internet, is its signaling system. Two broad approaches towards signaling can be identified: so-called hard-state and soft-state approaches. Despite the fundamental importance of signaling, our understanding of these approaches- their pros and cons and the circumstances in which they might best be employed- is mostly anecdotal (and occasionally religious). In this paper, we compare and contrast a variety of signaling approaches ranging from a “pure ” soft state, to soft-state approaches augmented with explicit state removal and/or reliable signaling, to a “pure ” hard state approach. We develop an analytic model that allows us to quantify state inconsistency in single- and multiple-hop signaling scenarios, and the “cost ” (both in terms of signaling overhead, and application-specific costs resulting from state inconsistency) associated with a given signaling approach and its parameters (e.g., state refresh and removal timers). Among the class of soft-state approaches, we find that a soft-state approach coupled with explicit removal substantially improves the degree of state consistency while introducing little additional signaling message overhead. The addition of reliable explicit setup/update/removal allows the soft-state approach to achieve comparable (and sometimes better) consistency than that of the hard-state approach. I.
Signaling for Internet Telephony
- in International Conference on Network Protocols (ICNP
, 1998
"... Internet telephony must offer the standard telephony services. However, the transition to Internet-based 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 ..."
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Cited by 41 (8 self)
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Internet telephony must offer the standard telephony services. However, the transition to Internet-based 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. 1. Introduction Internet ...
