Beyond VoIP Protocols by Olivier Hersent, Jean-Pierre Petit, and David
Gurle, Wiley, 2005.
This book covers what needs to be done to advance beyond a basic, end-to-end
Internet telephone system, particularly with respect to protocols.
Digital Telephony, third edition by John Bellamy, Wiley, 2000.
By the early 90s, about the same time the Internet was taking off as a public
utility, the American public telephone system completed its change-over from
analog to digital within the circle of central offices. This book describes
what the change-over from analog to digital entailed.
Digital Telephony Over Cable by D. R. Evans, Addison Wesley, 2001.
Covers PacketCable, the Cable Consortium’s set of standards specifying
a two-way digital communications system for cable TV systems.
Internet Telephony edited by Lee McKnight, William Lehr and David Clark,
MIT Press, 2001.
A somewhat risky book that tries to think its way into the future of
Internet-based telephony and communication systems more generally.
“What goes around comes around” is probably the most useful thing to be
thinking while reading this book.
Signaling and Switching for Packet Telephony by Matthew Stafford, Artech
What can be done once the bearer and control planes are separated into
Voice over IP by Uyless Black, Prentice Hall, 2002.
A good introductory book, reasonably complete and occasionally deep. It will
get you oriented in the VoIP landscape and sets you up to explore further.
Voice over IP Fundamentals by Jonathan Davidson and James Peters, Cisco
A book published, you will have noticed, by Cisco designed to make technical
managers comfortable and adept at constantly shoveling out more budget for
Cisco boxes one bigger than the ones they’ve already
VoIP Hacks by Ted Wallingford, O’Reilly, 2006.
A hodgpodge of tips & tools for Internet telephony.
Some of these papers are freely available, some require registration,
which you get automatically if you access the link from within
monmouth.edu domain. If you're not within
monmouth.edu domain and can't get there, you have to be a
member of the ACM or IEEE (depending on the paper) digital library.
Architecture for Residental Internet Telephony Service
by Christian Huitema, Jane Cameron, Petros Mouchtaris and Darek Smyk
in IEEE Internet Computing, May-June 1999 (v. 3, n. 3).
An internet-telephony archiecture should be able to handle millions of
end-points, integrate seamlessly with the public telenephone network (PTN)
including SS7 support, and be as reliable as the PTN. Given the
dissimilarites between the Internet and the PTN, the architecture should be
gateway-based, including a residential gateway, a trunking gateway, user
agents, and the usual media gateways.
Architecture for Secure VoIP and Collaboration Applications
by Dimitris Zisiadis, Spyros Kopsidas and Leandoros Tassiulas
in the Third International Workshop on Security, Privacy and Trust
in Pervasive and Ubiquitous Computing, 19 July 2007.
VoIP and collaboration Internet applications usually require registration in
a central user database and use either two bridged client-server connections
between the end users and the server or they allow direct client connections.
Biometric-based procedures followed by the VoIPSec (voice interactive
personalized security) protocol can provide end-to-end security for such
applications. This approach doesn’t need a trusted third-party
Anti-Vamming Trust Enforcement in Peer-to-Peer VoIP Networks
by Nilanjan Banerjee, Samir Saklikar and Subir Saha
in Proceedings of the 2006 International Conference on Wireless
Communications and Mobile Computing.
I send you a letter and seal it with a wax imprint. You trust the letter
came from me because the name and wax imprint match. Let my name be a bit
string n and the wax imprint be another bit string w with the
property that prefix(h(w), t) = prefix(n, t). prefix(b, n)
is the first (leftmost) n bits from the bit string b, h()
is a secure hash function, and t is a non-negative integer. Because
h() is impossible to invert, finding a wax imprint for which t
is large is expensive; wax imprints with large t values are more
trustworthy (in some sense) than wax imprints with small t values.
Using a public key from a public-key cryptosystem as my name provides
authentication by encoding the wax imprint with my private key.
Building Trustworthy Systems: Lessons from the PTN and Internet by Fred
Schneider, Steven Bellovin and Alan Inouye in IEEE Internet Computing, November-December
1999 (v. 3, n. 6).
The Internet and the public telephone network (PTN) have different ways of
being attacked; skills learned on one network don’t transfer to the
other. However, their increasing integration makes each an ingress for
attacks on the other. The PTN’s eroding monopoly status and the
Internet’s increasing commercialization gives rise to a cloud of
diverse, minimally-cooperative agents whose actions make matters worse.
What can go wrong is well known; what is to be done isn’t clear.
Critical VPN Security Analysis and New Approach for Securing VoIP
Communications over VPN Networks
by Wafaa Diab, Samir Tohme and Carole Bassil
in Proceedings of the 3rd ACM Workshop on Wireless Multimedia Networking
and Performance Modeling.
Many VoIP security attacks can be frustrated using encryption. VPN is a
standard mechansim for encrypting on the Internet, but is oriented toward
non-real-time data streams. VPN encryption for VoIP should support real-time
traffic using IP Security mechanisms and guarantee the performance and
quality of services without reducing the effective bandwidth.
by David Bryan and Bruce Lowekamp
in ACM Queue, March 2007 (v. 5, n. 2).
A peer-two-peer (p2p) overlay network responds naturally to network
connectivity and membership changes at the cost of introducing uncertainty
about network state. Hybrid p2p networks impose some structure - using, for
example, a distributed hash table - to reduce the uncertainty at a cost of
increasing the effort required to maintain the network. Session Initiation
Protocol (SIP) overlay networks are mostly distributed except for a few
centeralized services such as registration. Moving a SIP network to a p2p
network would make formally centeralized services unacceptably expensive, but
a hybrid p2p network may provide an appropriate trade-off between the ability
to react naturally to network-configuration changes and the cost of providing
formally centralized services.
Delay-Friendliness of TCP
by Eli Brosh, Salman Abdul Baset, Dan Rubenstein and Henning Schulzrinne
in Proceedings of the 2008 ACM SIGMETRICS International Conference on
Measurement and Modeling of Computer Systems.
Despite admonishions not to, many real-time Internet applications use TCP for
data transport. How does that work out for them? A Markov-chain model
validated by simulatons on networks shows that low packet-loss rates produce
small (< 1 sec.) TCP delays, as the loss rate increases the RTT should
decrease to compensate, and that large streams (500 Kb/s video) are more
effected than small streams (64 kb/s audio). Also, apart from the usual
parameter games (big window size, no Nagel, no byte counting, use SACK and so
on), splitting large packets into small ones may help the stream but may hurt
the network and using parallel streams helps muchly.
Economics of the Internet: Utility, Utilization, Pricing and Quality of Service
by Andrew Odlyzko, AT&T Research, 7 July 1998.
Can throwing bandwidth at the Internet solve congestion problems? Can it
solve congestion problems as efficiently and effectively as other approaches,
such as various quality of service (QoS) regimes? Many people say no, but
it’s not clear why that’s the correct answer.
Effect of Packet Dispersion on Voice Applications in IP networks by Hanoch
Levy and Haim Zlatorkrilov in IEEE/ACM Transactions on Networking, April 2006 (v. 14, n. 2).
Defines the noticeable packet loss (NPL) metric which weights packet
loss occurring close together over dispersed packet loss (that is, bursty
over Bernoulli loss) and then models how packet loss under dispersed packet
routing effects NPL. Packets are distributed among routes randomly,
cyclically, or round-robin. Route diversity does improve NPL, but the
assumptions used (particularly for independence and receive-side packet
handling) to carry the analysis gives one pause.
Enabling SIP-Based Sessions in Ad Hoc Networks
by Nilanjan Banerjee, Arup Acharya and Sajal Das
in Wireless Networks, August 2007 (v. 13, n. 4).
Session Initiation Protocol (SIP) servers running in the Internet have a
relatively stable infrastructure on which to build an overlay network for
endpoint discovery and session establishment. Ad hoc networks do not provide
a stable infrastructure and require extra techniques to support SIP-based
overlay networks. One technique, the loosely coupled approach, relies on the
underlying ad-hoc routing and provides endpoint discovery. Another
technique, the tightly coupled approach, includes session establishment by
defining a virtual topology among clusters of end-points. Simulations show
that tight coupling is better (has lower latency) in stable networks while
loose coupling is better in dynamic networks. In all cases the extra
structure provided by tight coupling has is less control overhead than does
End-To-End Arguments in System Design
by Jerome Saltzer, David Reed and David Clark
in ACM Transactions on Computing Systems, November 1984 (v. 2, n. 4).
What services should a network provide? The end-to-end argument
answers this question by assuming each service added to the network is
enormously expensive and requires showing that the enormous expense will be
amortized over all network users. If that totalizing amortization
can’t be carried out, the feature doesn’t belong in the
From POTS to PANS: A Commentary on the Evolution to Internet Telephony by
Christos Polyzois, Hal Purdy, Ping-Fai Yang, David Shrader, Henry Sinnreich,
François Ménard and Henning
Schulzrinne in IEEE Internet Computing,
May-June 1999 (v. 3, n. 3).
The Internet has a structure significantly different from that of the public
telephone network (PTN), both in the network and at the end-points. At least
initially, the Internet phone services will echo those of the PTN, raising
the question what should be brought over from the PTN and what should be
reconsidered anew. The PTN’s Intelligent Network infrastructure is
the most likely contact point for IP networks, both as a way to use
existing PTN services and functions and as a way to hook in new
Guaranteeing Multiple QoSs in Differentiated Services Internet
by Hoon Lee and Hyejin Kwon and Yoshiaki Nemoto
in Proceedings of the Seventh International Conference on Parallel and
An architecture to guarantee multiple Quality of Services (QoSs), including
the IETF’s Differentiated Service (DiffServ) architecture and the user
application’s requirements. A prioritized packet service scheme using
weighted round-robin in the core router supports weighted priority services
for the three IETF service classes: EF (Expedited forwarding), AF (Assured
forwarding) and DF (Default forwarding).
Holistic VoIP Intrusion Detection and Prevention System
by Mohamed Nassar, Saverio Niccolini, Radu State and Thilo Ewald
in Proceedings of the First International Conference on Principles,
Systems and Applications of IP Telecommunications, 2007.
often points out that several
flexible, lightweight security layers often combine to provide better
overall security than does a single, heavily armored bastion. Holistic VoIP
security illustrates Schneier’s point by using two layers to provide
VoIP security. The first layer is a VoIP honeypot to collect and analyze
data on attacks. The second layer is an event correlater that observes a
working VoIP system and flags operation sequences that seem suspicious.
Integrating Internet Telephony Services by Wenyu Jiang, Jonathan Lennox,
Sankaran Narayanan, Henning Schulzrinne, Kundan Singh and Xiaotao Wu
in IEEE Internet Computing, May-June 2002 (v. 6, n. 3).
Cinema (Columbia Internet extensible multimedia arechitecture) is a SIP-based
subsystem that hosts various multimedia facilities such as conferencing
(bridging), streaming media, unified voice messaging, and address resolution.
Cinema integrates with existing voice networks and end-points via SIP
proxies and gateways.
Integration of Call Signaling and Resource Management for IP Telephony by
Pawan Goyal, Albert Greenberg, Charles Kalmanek,
William Marshall, Partho Mishra, Doug Mortz and K. Ramkrishnan
in IEEE Internet Computing,
May-June 1999 (v. 3, n. 3).
An IP network usually has computing devices of varying power serving as
end-points and network nodes. A signaling architecture for such a network
should be distributed so work can be performed at the most appropriate
location and open so new services and old services re-implementations can
be easily added. Distribution requires scheduling to determine which
locations are appropriate and to dispatch work to those locations; QoS issues
— such as packet loss, delay, and jitter
— can be a first-cut driver for making scheduling
Modular Architecture for Providing Carrier-Grade SIP Telephony Services
by Hechmi Khlifi and Jean-Charles Grégorie in the
Third IEEE International Converence on Wireless and Mobile Cmmputing.
A modular, flexible and scalable architecture to provide mass-market
telephony services services in SIP environments. The architecture uses
Parlay, a standard, object-oriented and signaling protocol-neutral API,
and SIP to separate application logic and network function and, at the
network level, signaling and media processing.
Peer-to-Peer Internet Telephony Using SIP
by Kundan Singh and Henning Schulzrinne
in Proceedings of the International Workshop on Network and Operating
Systems Support for Digital Audio and Video, 13–14 June 2005, pages
Internet telephony (IT) networks embedded in the Internet have the usual tree
hierarchy structure. An alternative structure flattens IT subtrees (domains)
into a peer sets with no hierarchy. A flat domain should improve reliability
and change accommodation while making it harder to find resources. Session
Initiation Protocol servers in a flat hierarchy can run a peer-to-peer (P2P)
network protocol, such as Chord or Content-Addressable Network, to organize
themselves. However, typical P2P services are latency tolerant and exploit
resource replication while IT services are latency intolerant and
can’t easily replicate many resources (end users and databases, for
example). P2P security and economics models also match poorly with the
equivalent IT models.
Programming Internet Telephony Services by Jonathan Rosenberg, Jonathan
Lennox and Henning Schulzrinne in IEEE Internet Computing, May-June 1999 (v. 3, n. 3).
A control plane full of SIP servers can be induced to provide new services
using a CGI-like mechanism. New services are implemented as programs
independent of SIP servers and then invoked as independent processes by SIP
servers when the service is needed. A call-processing language,
circumscribed in its abilities to limit dangerous operations and to make it
statically checkable, makes it possible for end-users to implement custom
Providing Emergency Services in Internet Telephony by Henning Shulzrinne and
Knarig Arabshian in IEEE Internet Computing, May-June 2002 (v. 6, n. 3).
Emergency communications systems impose new requirements, such as universal
numbering, call routing, and caller number and location identification, as
well as the usual performance and reliability requirements on IP-based
voice-service networks. Replicating the emergency PSTN architecture is
(relatively) straightforward, but an IP network’s modular,
service-based structure allows for new architectures with better flexibility
Real-Time Voice Communication over the Internet Using Packet Path Diversity
by Yi Liang, Eckehard Steinbach and Bernd Girod
in Proceedings of the Ninth ACM International Conference on Multimedia,
The quality of real-time voice communication over best-effort networks is
mainly determined by the delay and loss characteristics observed along the
network path. Excessive playout buffering at the receiver is prohibitive and
significantly delayed packets have to be discarded and considered as late
loss. We propose to improve the tradeoff among delay, late loss rate, and
speech quality using multi-stream transmission of real-time voice over the
Internet, where multiple redundant descriptions of the voice stream are sent
over independent network paths. Scheduling the playout of the received voice
packets is based on a novel multi-stream adaptive playout scheduling
technique that uses a Lagrangian cost function to trade delay versus loss.
Experiments over the Internet suggest largely uncorrelated packet erasure and
delay jitter characteristics for different network paths which leads to a
noticeable path diversity gain. We observe significant reductions in mean
end-to-end latency and loss rates as well as improved speech quality when
compared to FEC protected single-path transmission at the same data rate. In
addition to our Internet measurements, we analyze the performance of the
proposed multi-path voice communication scheme using the ns network simulator
for different network topologies, including shared network links.
SCTP: A Proposed Standard for Robust Internet Data Transport
by Armando Caro, Jr., Janardhan Iyengar, Paul Amer, Sourabh Ladha, Gerard
Heinz, II and Keyur Shah
in IEEE Computer, November 2003 (v. 36, n. 11).
The Stream Control Transmission Protocol (SCTP) provides associations
between processes on hosts; each association contains one or more
unidirectional streams. SCTP provides flow- and congestion-controlled
reliable packet transport; each packet is mixture of control and data blocks.
SCTP end-points can straddle several ports on each host; set-up uses a
four-way handshake to avoid syn attacks and a three-way tear-down for speed
(and eliminating TCP’s half-close semantics).
Security Issues with the IP Multimedia Subsystem (IMS) by Michael Hunter,
Russ Clark and Frank Park in Workshop on Middleware for Next-generation
Converged Networks and Applications, Newport Beach, California, 26–30 November 2007.
The Internet Multimedia Subsystem (IMS) is designed to support convergent
services comprising voice and data. IMS security and related covers all the
usual suspects (QoS, billing, services, regulation, security) from the
providers’ and users’ perspectives.
Apart from a new, more complex architecture, IMS-relevant consideration of
these areas will be familiar to those with experience in other areas of
Internet-based subsystem design.
Security Patterns for Voice over IP Networks by Eduardo Fernandez and
Juan Pelaez and Maria Larrondo-Petrie in Proceedings of the International
Multi-Conference on Computing in the Global Information Technology,
4–9 March, 2007.
The grand convergence of voice, video and data on VoIP networks is a source
of great hope, but also a source of security concerns do to the lack of
isolation between the bit streams. Various system structures, described as
software patterns can re-establish isolation to improve security. The
patterns involve encryption, network segmentation, tunneling, and
Session Initiation Protocol: Internet-Centric Signaling by Hennig
Schulzrinne and Jonathan Rosenberg in IEEE Communications, October 2000
(v. 38, n. 10).
The Session Initiation Protocol (SIP) provides signaling and control for
multimedia services. SIP locates resources based on a location-independent
name and negotiates session characteristics. It can be used for Internet
telephony and conferencing, instant messaging, event notification, and the
control of networked devices. SIP is a typical IETF protocol: text-based,
line-oriented, request-response. Designed to be extensible, SIP has been
extended in several ways to define new services (instance messaging, for
example) and features (authentication, for example).
SIP-Based Conference Control Framework
by Petri Koskelainen, Henning Schulzrinne and Xiaotao Wu
in Proceedings of the 12th International Workshop on Network and
Operating Systems Support for Digital Audio and Video.
Conference services in Internet-telephony (IT) systems should be implemented
in a way consistent with IT to reap the benefits of such systems. SIP-based
coordination using SOAP provide the mechanisms for conference and floor
control. Central SIP servers and unicast should be good enough for small
conferences, but larger conferences probably require distributed servers or
multicast or both.
SOVoIP: Middleware for Universal VoIP Connectivity
by M. J. Arif and S. Karunasekera and S. Kulkarni
in 8th ACM/IFIP/USENIX International Conference on Middleware.
VoIP has a number of protocols that don’t interoperate, but instead
are coordinated by protocols such as SIP or H323. For some reason, SIP or
H.323 don’t look enough like middleware, so maybe they can be replaced
(or suplimented, it isn’t clear) by CORBA or web services. Naturally
CORBA is right out, because of its firewall difficulty and performamce,
leaving web services in the form of Service Oriented VoIP (SOVoIP). Just to
make sure, SOVoIP performs better than CORBA, but no comparisons are made
with SIP or H.232.
Terminating Telephony Services on the Internet by
Vijay Gurbani and Xian-He Sun in IEEE/ACM Transactions on Networking, August
2004 (v. 12, n. 4).
How to originate a service in the telephone network and terminate it in an
Internet-based network using standard protocols (SIP, HTTP, XML) and a
publish-subscribe architecture. The desire to avoid middleware is admirable,
but requiring direct access to signaling is troubling. It’s also
unclear whether the same architecture can apply in the Internet-to-telephone
Time Synchronization for VoIP Quality of Service
by Hugh Melvin and Liam Murphy
in IEEE Internet Computing,
May-June 2002 (v. 6, n. 3).
Effectively handling time-sensitive voice playout over the Internet requires
good and stable information about end-to-end delays. Relatively simple
estimation at the receiver’s end works well as long as the estimates
don’t drift too rapidly. Time synchronization via GPS provides a
uniform, stable time signal end-points can use to produce accurate, stable
Towards a new Security Architecture for Telephony
by Carole Bassil, Ahmed Serhrouchni and Nicolas Rouhana
in Proceedings of the International Conference on Networking,
International Conference on Systems and International Conference on Mobile
Communications and Learning Technologies (ICNICONSMCL ’06).
The telephone and VoIP networks place different emphasis in their security
policies and use different machanisms to acheive their policies. This
difference is yet another gap that has to be bridged in the networks’
grand convergence. However, rather than using gateways to translate between
the security mechanisms, a shim layer in each network protocol stack would
allow each security mechanism to be translated to a common mechanism
providing a secure end-to-end voice communication.
Tussle in Cyberspace: Defining Tomorrow’s Internet by
David Clark, John Wroclawski, Karen Sollins and Robert Braden in IEEE/ACM Transactions on Networking, June
2005 (v. 13, n. 3).
A tussle is a clash of interests among competing parties in a system.
The Internet was designed and implemented in a relatively tussle-free
environment; however, the Internet’s current popularity and importance
has increased the number and diversity of competing parties and greatly
increased the number of tussles, making the original design principles less
useful then they once were. New design principles should recognize and
identify places where tussles may occur and support late binding to allow a
range of possible resolutions.
Ubiquitous Computing using SIP
by Stefan Berger, Henning Schulzrinne, Stylianos Sidiroglou and Xiaotao Wu
in Proceedings of the 13th International Workshop on Network and
Operating Systems Support for Digital Audio and Video.
The Session Initiation Protocol (SIP) is an open, extensible, distributed,
request-response infrastructure. Extending a SIP-based communication system
with user-location information allows for services that follow you around and
customize themselves to your location. Such an extension requires a
subsystem for discovering user location, a subsystem for managing location
information, and a subsystem for reacting to location state.
Unified Communications with SIP
by Martin Steinmann
in ACM Queue, March 2007 (v. 5, n. 2).
Proprietary PBXs are disappearing because standard and open-source
Internet-telephony software, such as SIP, can provide similar services more
flexibility and less cost, and are easy to extend to provide new services.
Voice Over IP Service Architecture for Integrated Communications by Daniele
Rizzetto and Claudio Catania in IEEE Internet Computing, May-June 1999 (v.3, n. 3).
The unification of voice and data traffic in the Internet overshadows an
increasing separation between the control and data parts of the network.
Emphasizing the control-data separation can make it simpler to efficiently
implment advanced services, as well as well as isolate each part from
technological change in the other part. A service architecture providing an
abstract API to control network preserves the advantages of separated control
VoIP Security and Privacy Threat Taxonomy,
VOIPSA, 24 October 2005.
All (most? some? a few?) of the goblins that could get you if you
don’t watch out.
VoIP Security: Not an Afterthought by Douglas Sicker and Tom Lookabaugh in
ACM Queue, September 2004 (v. 2, n. 6).
The things that make VoIP interesting and important —
distributed operation, flexibility, openness — also makes
it hard to secure. One advantage is an Internet base, which come with
existing relevant security work and research.
VoIP: What is it Good For? by Sudhir Ahuja and Robert Ensor in ACM Queue,
September 2004 (v. 2, n. 6).
A brief, high-level comparison between service implementation in the PSTN and
over VoIP networks, mostly to the favor of VoIP networks. Recognizes the
importance of service development the the future of VoIP networks, but then
presents lame examples (click-to-dial web page links, persistent chat).
You Don’t Know Jack About VoIP by Phil Sherburne and
Cary Fitzgerald in ACM Queue, September 2004 (v. 2, n. 6).
Voice over Internet shows great promise due to network flexibility and
openess, but also presents great challenge given the service requirements for
good quality voice traffic, as well as management and security requirements.
This page last modified on 14 August 2008.