SIP and Internet Telephony: Papers, Books and Talks

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SIP IMS Specifications for Dummies eBook
Uri Baniel
Self-published by SIPKnowledge, 2007

The IMS: IP Multimedia Concepts and Services
Miikka Poikselka, Aki Niemi, Hisham Khartabil, Georg Mayer
ISBN 0470019069

The 3G IP Multimedia Subsystem (IMS): Merging the Internet and the Cellular Worlds
Gonzalo Camarillo, Miguel-Angel García-Martín
ISBN 0470018186

SIP, TCP/IP und Telekommunikationsnetze Anforderungen - Protokolle - Architekturen
Ulrich Trick and Frank Weber
2004. 375 Seiten, mit CD-ROM; 49,80 Euro; ISBN 3-486-27529-1 (in German)

Voice-Enabling the Data Network: H.323, MGCP, SIP, Q.S, SLAs, and Security
James Durkin, Cisco Press. 1587050145

Internet Communication Using SIP: Delivering VoIP and Multimedia Services with Session Initiation Protocol
Henry Sinnreich and Alan B. Johnston, Wiley, 2001. ISBN 0-471-41399-2.

SIP Demystified
Gonzalo Camarillo, McGraw-Hill Professional Book Group, 2001. ISBN 0-07137340-3.

SIP: Understanding the Session Initiation Protocol
Alan B. Johnston, Artech House, second edition, 2004. ISBN 1-58053-655-7

IP Telephony
Olivier Hersent, David Gurle, Jean-Pierre Petit. Addison-Wesley, 2000. ISBN 0-201-61910-5.

Big Book of IP Telephony RFCs
Peter Loshin. Morgan Kaufmann, 2001. ISBN 0-124-558550


SIP - The Key to VoIP
Handouts from 2-day seminar (paid)
SIP servers
Whitepaper by Radvision, explaining basic SIP operation.
SIP and the new network communications model (2003)
The purpose of this document is to provide an overview of SIP-the Session Initiation Protocol-and explain how SIP is reshaping the communications landscape. SIP enables an Internet-based architecture used to manage communication "sessions" over IP networks, enabling converged voice and multimedia services while promoting natural communications between people, not devices. This document will also review the significant business implications of this new protocol.
SIP Specifications and the Java Platforms
White paper by Phelim O'Doherty, 2003, "identify the specifications to the Session Initiation Protocol (SIP) defined through the Java Community Process (JCP)".
Slides for SIP course, by Charles Baker (with other contributors)
Session Initiation Protocol (SIP) Tutorial
Slides, by Jiri Kuthan and Dorgham Sisalem
SIP: Designed for Interoperability
Jonathan Rosenberg
Internet Telephony, October 2000.
The Session Initiation Protocol (SIP): A Key Component for Internet Telephony, Computer Telephony, June 2000 (Vol. 8, Issue 6).
Jonathan Rosenberg and Richard Shockey

Last month, we got our first look inside the SIP standard for signaling communications services on the Intenet and emerging SIP products. This month, we've gone to principal source for a more thorough primer.

Internet Telephony: architecture and protocols - an IETF perspective
Henning Schulzrinne and Jonathan Rosenberg
Computer Networks, February 11, 1999. Vol. 31, No. 3.

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.

The Session Initiation Protocol (SIP)
Henning Schulzrinne
Slides on Internet telephony and multimedia, last updated August 2001.


Lamont-Doherty Earth Observatory

case study for using SIP on a ship

General Reliability and Security Framework for VoIP Infrastructures
Deliverable for SNOCER project, a research project supported within the Sixth Framework Programme of the EU Commission; 2006

Remove vulnerability from SIP-based VoIP networks -- Part (May 22, 2006), Part 2 (May 26, 2006)
Vinay Rao
Network Systems DesignLine, May 2006

vCert white papers on requirements for SIP phones, May 2003.

Towards Junking the PBX: Deploying IP Telephony
Wenyu Jiang, Jonathan Lennox, Henning Schulzrinne and Kundan Singh
NOSSDAV 2001, Port Jefferson, New York, June 2001. slides

The future of converged networks
Jonathan Rosenberg
Communications Solutions, March 2001

A Service Platform for Internet-Telecom Services using SIP
S. Bessler, A. V. Nisanyan, K. Peterbauer, R. Pailer, J. Stadler
Smartnet 2000

This paper proposes the introduction of a Service Platform for the creation, execution and management of multimedia services in heterogeneous networks. Examining the business-roles, the actors in the Service Platform are identified. Furthermore several common building blocks for developing services for the Internet are described and a brief overview of some modern technologies for an object-oriented, component-based, distributed platform for multimedia services is given. The Session Initiation Protocol (SIP) has been identified as very useful to implement all functions according to the multimedia part of the Platform. The usage of the PARLAY API as an open interface between the Platform Services and SIP, the PSTN or the mobile network provides a lot of additional advantages. The interworking between SIP and PARLAY is shown in a call-routing example. Furthermore the call-setup for a multimedia (e.g. video) conference is explained. This should demonstrate the usefulness and the ability of this protocol for introducing a session concept. Finally an outlook of open research topics regarding this concept is given as well as a short overview of related work.

Application-Layer Mobility using SIP
Henning Schulzrinne and Elin Wedlund
Mobile Computing and Communications Review (MC2R), Volume 4, Number 3, July 2000.

Supporting mobile Internet multimedia applications requires more than just the ability to maintain connectivity across subnet changes. We describe how the Session Initiation Protocol (SIP) can help provide terminal, personal, session and service mobility to applications ranging from Internet telephony to presence and instant messaging. We also briefly discuss application-layer mobility for streaming multimedia applications initiated by RTSP.

Clients Thick, Clients Thin
Tony Rybscynsky, Nortel Networks, Enterprise Solutions. Communications Solutions Magazine, Volume 1, July 2000.

From where does IP telephony derive its power?New applications. Some of these applications bring the human touch to e-Business. Others serve the collaboration needs of an increasingly distributed workforce.And yet others signal opportunities for enhanced connectivity,specifically,a broader array of connectivity options that take advantage of Internet ubiquity and LAN plug-and-play capabilities.

Presence: The Best Thing That Ever Happened To Voice
Jonathan Rosenberg
Computer Telephony, October 2000.

Forget Caller ID. A new group of 'presence' technologies and standards will let applications know where you are, what you're doing, and what kind of communications you're prepared to receive.

Succession Solutions Integration of Session Initiation Protocol
Nortel Networks. Technical Brief. September 2000.

With the emerging Session Initiation Protocol (SIP), you can deliver a range of personalized, multi- media-rich services anywhere, anytime. Deregulation, subscriber mobility, and business communication outsourcing have driven competition among service providers to a fever pitch. To retain the existing subscriber base and expand into new markets, successful service providers are looking for ways to deliver personalized, differentiated, real-time multimedia services, such as collaborative meetings and subscriber-initiated service management. By incorporating SIP capabilities, Succession solutions extend to service providers a key competitive edge by increasing end-user productivity, expanding subscriber mobility, and enhancing interactive communications.

Unified Messaging using SIP and RTSP
Kundan Singh and Henning Schulzrinne
IP Telecom Services Workshop, Sept. 11, 2000. Atlanta, Georgia.

Traditional answering machines and voice mail services are closed systems, tightly coupled to a single end system, the local PBX or local exchange carrier. Even simple services, such as forwarding voice mail to another user outside the local system, are hard to provide. With the advent of Internet telephony, we need to provide voice and video mail services. This also offers the opportunity to address some of the shortcomings of existing voice mail systems. We list general requirements for a multimedia mail system for Internet telephony. We then propose an architecture using SIP (Session Initiation Protocol) and RTSP (Real-Time Streaming Protocol) and compare various alternative approaches to solving call forwarding, reclaiming and retrieval of messages. We also briefly describe our prototype implementation.",

Building Voice over IP
Philip Carden. Network Computing, May 2000.

Outlines modes of Internet telephony and offers a cursory comparison of SIP and H.323.

Feature Interaction in Internet Telephony
Jonathan Lennox and Henning Schulzrinne. Proceedings of the Sixth International Workshop on Feature Interactions in Telecommunications and Software Systems, May 2000.

While Internet telephony aims to provide services at least equal to traditional telephony, the architecture of Internet telephony is sufficiently different to make it necessary to revisit the issue of feature interaction in this context. While many basic feature interaction problems remain the same, Internet telephony adds additional complications. Complications arise since functionality tends to be more distributed, users can program the behavior of end systems and signaling systems, the distinction between end systems and network equipment largely vanishes and the trust model implicit in the PSTN architecture no longer holds. On the other hand, Internet telephony makes end point addresses plentiful and its signaling makes it easy to specify in detail the desired network behavior. Many techniques for resolving interactions in the PSTN are no longer easily applied, but several new techniques, explicitness, authentication, and verification testing, become possible in the Internet environment.

Integrating presence with multi-media communications
Russell Bennett and Jonathan Rosenberg. dynamicsoft white paper. May 2000.

Until recently, the compelling value proposition for IP-based voice communications was toll-bypass. Despite Quality of Service (QoS) issues, people have been willing to make IP-based phone calls because it was free and it was fun. However, the cost differentials are rapidly eroding. As a result, service providers must find a new value proposition for IP communications. We believe that this value proposition comprises a range of new services that take advantage of other IP applications, such as web, email, instant messaging and most importantly, Presence. Blending these applications with voice means that a whole new set of features and communications experiences are enabled for consumers. Presence, in particular, can significantly enhance communications services for consumers. Unfortunately, despite the popularity of Presence and instant messaging systems on the Internet, there is no open standard, and insufficient support for multimedia. We believe that the Session Initiation Protocol (SIP), already an integral part of communications services on the Internet, can serve as a strong foundation for building an open, scalable, secure, multimedia-enabled Presence protocol.

Parched For Services? Here, Try A SIP.
Jonathan Rosenberg
Communications Solutions, May 2000.

A Comparison of H.323v4 and SIP
Nortel Networks
3GPP S2, Tokyo Japan, S2-000505

This contribution compares and contrasts SIP to H.323v4 to help aid operators and vendors in the selection of a single least common denominator control protocol for the ps "domain" or perhaps more appropriately "plane" of UMTS Release 2000. The format anticipates the concerns, in the form of questions, which may arise from 3GPP members.

Personal mobility for multimedia services in the Internet PDF
Henning Schulzrinne
European Workshop on Interactive Distributed Multimedia Systems and Services, (Berlin, Germany), Mar. 1996.

Personal mobility is one of the goals of Universal Personal Telecommunications (UPT) being specified for future deployment. Most current efforts focus on telephony, with SS7 signaling. However, many of the same goals can be accomplished for multimedia services, by using existing Internet protocols. We describe a multimedia call/conference setup protocol that provides personal videophone addresses, independent of the workstation a called party might be using at the time. The system is set up to use the existing Internet email address as a videophone address. Location and call handling information is kept at the subscriber's home site for improved access and privacy.

Comparison of H.323 and SIP for IP Telephony Signaling
Ismail Dalgic and Hanlin Fang
Proc. of Photonics East, Boston, Massachusetts, September 20-22, 1999.

Two standards currently compete for the dominance of IP telephony signaling: the H.323 protocol suite by ITU-T, and the Session Initiation Protocol (SIP) by IETF. Both of these signaling protocols provide mechanisms for call establishment and teardown, call control and supplementary services, and capability exchange. We investigate and compare these two protocols in terms of Functionality, Quality of Service (QoS), Scalability, Flexibility, Interoperability, and Ease of Implementation. For fairness of comparison, we consider similar scenarios for both protocols. In particular, we focus on scenarios that involve a gatekeeper for H.323, and a Proxy/Redirect server for SIP. The reason is that medium-to-large IP Telephony systems are not manageable without a gatekeeper or proxy server. We consider all three versions of H.323. In terms of functionality and services that can be supported, H.323 version 2 and SIP are very similar. However, supplementary services in H.323 are more rigorously defined, and therefore fewer interoperability issues are expected among its implementations. Furthermore, H.323 has taken more steps to ensure compatibility among its different versions, and to interoperate with PSTN. The two protocols are comparable in their QoS support (similar call setup delays, no support for resource reservation or class of service (CoS) setting), but H.323 version 3 will allow signaling of the requested CoS. SIP's primary advantages are (i) flexibility to add new features, and (ii) relative ease of implementation and debugging. Finally, we note that H.323 and SIP are improving themselves by learning from each other, and the differences between them are diminishing with each new version.

Case study: multimedia conference control in a packet-switched teleconferencing system
Eve M. Schooler
Journal of Internetworking: Research and Experience, vol. 4, pp. 99-120, June 1993. ISI reprint series ISI/RS-93-359.

MMCC, the multimedia conference control program, is a window-based tool for conference management. It serves as an application interface to the ISI/BBN teleconferencing system, where it is used not only to orchestrate multisite conferences, but also to provide local and remote audio and video control, and to interact with other conference-oriented tools that support shared workspaces. The motivation for this paper is to document the design, operation and continued evolution of MMCC. After presenting the context for this work, we provide a discussion of MMCC's peer-to-peer model of communication and an overview of its connection control protocol. Issues are also raised about heterogeneity, robust services, scalability and the impact of conferencing over the Internet. A description of the system's regular use offers insight into the feasibility of the architecture. Finally, future directions for research in multimedia conference control are presented.

SIP, NAT and Firewalls
Fredrik Thernelius. Master's thesis, Department of Teleinformatics, Kungl Tekniska Hgskolan, May 2000.

The work presented in this Master's Thesis is an examination of how the SIP signaling, which occurs when a so called IP Telephony session is set up, will be able to traverse firewalls. It is necessary to solve the problems/issues that SIP brings about when the SIP messages traverse firewalls if this protocol ever will gain popularity. In order to set up those data streams needed for transporting the sound in an IP telephony session the client enters his IP address and a port number in the SDP part of the SIP message to tell the other party where he should sent his audio data. Here is where problems occurs with the firewall. It needs to understand and interpret what the SIP message says to be able to set up rules for allowing traffic to pass through the firewall to these addresses. The problem is extended by the fact that it is common today to use "private addresses" on the LAN. These addresses are not allowed to exist on the Internet and thus the firewall software must remove this address and replace it with an address that is allowed on the Internet. A Network Address Translator (NAT) in the firewall normally does this together with Application Level Gateways (ALGs). The work of this Master's Thesis has been focused around analyzing the above mentioned problems with SIP and Firewalls and then using this as input designing a prototype of an Application Level Gateway for SIP, which could be used together with perhaps a Linux firewall.

SIP (RFC 2543), an Implementation for Marratech Pro, May 2000.
Petter Tiilikainen

This Master Thesis discusses SIP, Session Initiation Protocol, which provides services for user location, determination of user availability and media negotiation for the setup of subsequent multimedia sessions. Also discussed is the author's implementations of a few SIP components, and how they could be used together with Marratech Pro, an application for multimedia conferencing developed by Marratech AB. SIP is then examined in terms of its relation to H.323, another protocol for setup and management of multimedia sessions, and the possibilities for SIP to coexist with this protocol.

Predicting Internet Telephony Call Setup Delay
Tony Eyers and Henning Schulzrinne
IPTel 2000 (First IP Telephony Workshop), Berlin, April 2000.

Internet telephony has been the focus of much recent effort by ITU and IETF standards bodies, with initial, albeit small-scale deployment in progress. While Internet telephony voice quality has been studied, call setup delay has received little attention. This paper outlines a simulation study of Internet Telephony Call Setup delay, based on UDP delay/loss traces. The focus is signaling transport delay, and the variations arising from packet loss and associated retransmissions. Of particular interest are the differences arising from H.323 signaling, which uses TCP, and SIP, which can use UDP with additional error recovery. Results show that during high error periods, H.323 call setup delay significantly exceeds that of SIP. We also consider PSTN/Internet telephony interworking, and show that high blocking rates are likely if either H.323 or SIP are used across the public Internet.

An implementation of the Internet Call Waiting service using SIP
Inmaculada Espigares del Pozo.
Master's thesis at Helsinki University of Technology and Polytechnic University of Valencia. December 1999.

We have studied the SIP for the purpose of evaluating it and to make an implementation of a new service, the Internet Call Waiting (ICW). It is a useful solution for the calls that otherwise would be lost when the line is busy and also for rejecting undesirable incoming calls. On the other hand, it is a way of not wasting network resources and contributing to call completion. Thus, pop-up dialogue boxes are presented to make it simpler and easier to the user whose satisfaction is always an important objective for an IP based service. For service implementation, as the main tool, we have used the XML language. XML is considered one of the best languages for describing complex data relationships. We have also chosen XML because it is easily extended, flexible and because it has a text-based syntax. The complete project consists of a JAVA program that implements an UAS/UAC running in a PC and also an extension (embedding the XML parser) of the SIP server written in C borrowed from Columbia University to handle the scripts written in XML defining the service required by the users. In conclusion, we have tried to use the most efficient tools and mechanisms to complete this work as we consider that time and money are resources to take into account when developing the services of the new era.

An architecture of Conference Control Functions
Nadia Kausar and Jon Crowcroft
Proc. of Photonics East, Boston, Massachusetts, September 20-22, 1999.

Conference control is an integral part in many-to-many communications that is used to manage and co-ordinate multiple users in conferences. There are different types of conferences which require different types of control. Some of the features of conference control may be user invoked while others are for internal management of a conference. In recent years, ITU (International Telecommunication Union) and IETF (Internet Engineering Task Force) have standardised two main models of conferencing, each system providing a set of conference control functionalities that are not easily provided in the other one. This paper analyses the main activities appropriate for different types of conferences and presents an architecture for conference control called GCCP (Generic Conference Control Protocol). GCCP interworks different types of conferencing and provides a set of conference control functions that can be invoked by users directly. As an example of interworking, interoperation of IETF's SIP and ITU's H.323 call control functions have been examined here. This paper shows that a careful analysis of a conferencing architecture can provide a set of control functions essential for any group communication model that can be extensible if needed.

SIP Telephony Gateway on DTM
Mattias Eriksson and Lars Lundstedt
Bachelor Thesis. The Royal Institute of Technology, Haninge. June 1999.

SIP telephony Gateway on DTM, Mattias Eriksson and Lars Lundstedt, The Royal Institute of Science, KTH-Haninge, Sweden and AU-system AB, Liljeholmen June 99} The future of IP-telephony looks bright. Many companies now have realized the possibilities with this technique. The benefits of transport voice and data on the same network are probably the main reason. The setup is a test network, with "simple" SIP user agent and "simple" SIP Server/Gateway implementations, connected to PSTN with a Dialogic D41/ESC board. A part of the network is DTM(Dynaminc synchronous Transfer Mode) technology with dynamic bandwidth allocation. The thesis also gives an introduction to SIP and DTM.

Compaq CustomSystems Gatekeeper Implementation
Michel A. Maddux
Compaq DIGITAL technical whitepaper, 1999.

Mobility Support using SIP
Elin Wedlund and Henning Schulzrinne
Second ACM/IEEE International Conference on Wireless and Mobile Multimedia (WoWMoM'99), Seattle, Washington, August, 1999.

Enabling mobility in IP networks is an important issue for making use of the many light-weight devices appearing at the market. The IP mobility support being standardized in the IETF uses tunnelling of IP packets from a Home Agent to a Foreign Agent to make the mobility transparent to the higher layer. There are a number of problems associated with Mobile IP, such as triangular routing, each host needing a home IP address, tunnelling management, etc. In this paper, we propose to use mobility support in the application layer protocol SIP where applicable, in order to support real-time communication in a more efficient way.

Internet telephony protocols
Linden deCarmo
Dr. Dobbs Journal, July 1999.

Linden examines the strengths and weaknesses of SIP and H.323, the two dominant "Voice over the Internet" protocols. He also takes a look at a new challenger -- the Media Gateway Control Protocol.

True Number Portability and Advanced Call Screening in a SIP-Based IP Telephony System
Ismail Dalgic, Michael Borella, Rick Dean, Jacek Grabiec, Jerry Mahler, Guido Schuster, and Ikhlaq Sidhu
IEEE Communications Magazine, Vol. 37, No. 7, July 1999.

Custom local area signaling service features offered in the PSTN have certain limitations due to the closed nature of PSTN network signaling. The adoption of telephony over IP (IP telephony) will enable a new paradigm of services and features that are not possible to implement in today's PSTN. This is especially the case for services that make use of personal, trusted information, which can be provided by a user's personal digital assistant. In this article we demonstrate how personal information can be coupled with an IP telephony service to provide user-customized call handling by the network. In particular, we describe a demonstration architecture that includes Ethernet-attached phones running SIP, with an interface to synchronize with PDAs that supply personal information. The proposed architecture is quite flexible; it can support enhanced versions of the current PSTN and private branch exchange services, in addition to many new features and services. We describe true number portability and advanced call screening as examples of new services in a hybrid PSTN/IP telephony environment.

Interaction of Call Setup and Resource Reservation Protocols in Internet Telephony
Henning Schulzrinne, Jonathan Rosenberg and Jonathan Lennox
Technical report, June 1999.

In the Internet, call signaling, security association and resource reservation are handled by separate protocols and likely traverse different paths. However, for reliable service, the three functions may need to be coupled during call setup. We describe and compare several approaches to coupling, based on either single-phase setup or two-phase setup mechanisms. Our discussion is based on the Session Initiation Protocol (SIP), but also applies to other signaling protocols with similar properties.

Programming Internet Telephony Services
Jonathan Rosenberg, Jonathan Lennox and Henning Schulzrinne
IEEE Network Magazine, Vol. 13, No. 3, May/June 1999, pg. 42-49 and IEEE Internet Computing, Vol. 3, No. 3, pg. 63-72, May/June 1999.

Programming new Internet telephony services requires decisions regarding such things as where the code executes and how it interfaces with network protocols. The paper describes SIP cgi-bin and the Call Processing Language (CPL).

Programming Internet Telephony Services PDF
Jonathan Rosenberg, Jonathan Lennox and Henning Schulzrinne
Columbia University Computer Science Technical Report CUCS-010-99, March 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 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.

The IETF Internet Telephony Architecture and Protocols
IEEE Network Magazine, Vol. 13, No. 3, May/June 1999, pg. 18-23.

Protocols that provide a partial solution for interworking Internet telephony and traditional circuit-switched telephony are presented. The article provides a concise introduction to SIP.
A SIP of IP-telephony
Fredrik Fingal and Patrik Gustavsson
Department of Communication Systems, Lund Institute of Technology, Lund University and Sigma Exallon Systems AB, Malm, February 1999.

There are two different protocols competing in the Internet telephony world today, the Session Initiation Protocol (SIP) emerged from IETF and the H.323 emerged from ITU. The IP-telephony market is growing and will most likely affect the traditional circuit-switched telephony business in the future. We have studied the Session Initiation Protocol in the purpose of evaluating it and to make a small implementation of a client and a server. The tools Rational Rose and Java was used for the implementation. We have also compared it against the H.323 and tried to give a hint of what the future holds.

Implementing Intelligent Network Services with the Session Initiation Protocol PDF
Jonathan Lennox, Henning Schulzrinne and Thomas F. La Porta
Columbia University Computer Science Technical Report CUCS-002-99, January 1999.

Internet telephony is receiving increasing interest as an alternative to traditional telephone networks. This article shows how the IETF's Session Initiation Protocol (SIP) can be used to perform the services of traditional Intelligent Network protocols, as well as additional services.

The Session Initiation Protocol: Providing Advanced Telephony Services Across the Internet PostScript PostScript
Henning Schulzrinne and Jonathan Rosenberg
Bell Labs Technical Journal, Vol. 3, November/December 1998.
During the past few years, Internet telephony has evolved from a toy for the technically savvy to a technology that, in the not too distant future, may replace the existing circuit-switched telephone network. Supporting the widespread use of Internet telephony requires a host of standardized protocols to ensure quality of service (QoS), transport audio and video data, provide directory services, and enable signaling. Signaling protocols are of particular interest because they are the basis for advanced services such as mobility, universal numbers, multiparty conferencing, voice mail, and automatic call distribution. Two signaling protocols have emerged to fill this need: the ITU-T H.323 suite of protocols and session initiation protocol (SIP), developed by the Internet Engineering Task Force (IETF). In this paper we examine how SIP is used in Internet telephony. We present an overview of the protocol and its architecture, and describe how it can be used to provide a number of advanced services. Our discussion of some of SIP's strengthsits simplicity, scalability, extensibility, and modularityalso analyzes why these are critical components for an IP telephony signaling protocol. SIP will prove to be a valuable tool, not just for end-to-end IP services, but also for controlling existing phone services.

IP Telephony Gateways
Gonzalo Camarillo (Ericsson Telecom AB, Department of Teleinformatics, KTH)
Master's thesis, Nov. 1998
There are two different approaches to provide telephone services at present: to use the traditional switched network and to use the Internet. Both approaches employ different ways to establish connections, transmit the voice and terminate calls. This study focuses on the establishment of connections and the release of them. Different protocols are analysed and finally SIP and SS7 are described. The possible compatibility between them and the mapping between both message formats are also analysed. Features of both networks are described, and the functions of a gateway between them are outlined.
Internet Telephony: Architecture and Protocols -- an IETF Perspective
Henning Schulzrinne and Jonathan Rosenberg
to appear in Computer Networks and ISDN Systems

Internet telephony offers the opportunity to design a global multimedia communications systems that may eventually replace the existing telephony infrastructure, without being encumbered by the legacy of a century-old technology. 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 RTSP for control of streaming media, and WASRV for location of telephony gateways.
A Comparison of SIP and H.323 for Internet Telephony PDF
Henning Schulzrinne and Jonathan Rosenberg
Network and Operating System Support for Digital Audio and Video (NOSSDAV), (Cambridge, England), July 1998.

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 features, services, potential for future growth, and implementability.
Signaling for Internet Telephony PDF
Henning Schulzrinne and Jonathan Rosenberg
Columbia University Technical Report CUCS-005-98; submitted for publication, January 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 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.

A comprehensive multimedia control architecture for the Internet PDF
Henning Schulzrinne
Proc. International Workshop on Network and Operating System Support for Digital Audio and Video (NOSSDAV), (St. Louis, Missouri), May 1997.
The Internet and intranets have been used to deliver continuous media, both stored and interactive, for a number of years. Most of the attention has focused on providing guaranteed quality of service (RSVP) and end-to-end data transport (RTP), with every application using its own control protocol. In this paper, we describe a control architecture that supports most standard advanced telephony features and allows to integrate stored and interactive multimedia. The protocol re-uses much of the ``infrastructure'' of HTTP, including its security and proxy mechanisms. The architecture is instantiated by two related, but independent protocols: the Session Initiation Protocol (SIP) for inviting participants to a multimedia session and the Real-Time Stream Protocol (RTSP) to control playback and recording for stored continuous media.
Signaling for internet telephony services
Henning Schulzrinne
Proc. of Opensig'96, (New York, New York), Oct. 1996.

See also papers on Internet multimedia and resource reservation.

Last updated by Henning Schulzrinne