Network Systems DesignLine, May 2006
NOSSDAV 2001, Port Jefferson, New York, June 2001.
slides
Communications Solutions, March 2001
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.
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.
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.
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.
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.
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.",
Outlines modes of Internet telephony and offers a cursory comparison of
SIP and H.323.
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.
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.
Communications Solutions, May 2000.
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.
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.
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.
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.
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.
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.
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.
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.
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.
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 DIGITAL technical whitepaper, 1999.
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.
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.
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.
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.
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).
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.
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.
Department of Communication
Systems, Lund Institute of Technology, Lund University and Sigma
Exallon Systems AB, Malmö,
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.
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.
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
strengths—its simplicity, scalability, extensibility, and
modularity—also 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.
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.
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.
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.
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.
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.
Proc. of Opensig'96, (New York, New York), Oct. 1996.