Session Initiation Protocol (SIP) has become a strong force shaping the IP

telephony and unified communications (UC) industries, and is quickly

becoming the request-response protocol of choice among IT pros. SIP's
beauty lies in its simplicity, which allows unlimited scalability and
performance in different architectures and environments. Over the last few
years, the VoIP community has established SIP as its primary choice for
signaling. SIP's ability to establish sessions in an IP network for services
like instant messaging (IM) and collaboration tools has also allowed it to
become an integral part of unified communications.

This guide details the ins and outs of SIP, from basic architecture to
advanced troubleshooting and security.

TABLE OF CONTENTS

SIP basics
VoIP signaling protocols
How does SIP work?
Security and troubleshooting
SIP hardware
SIP in the enterprise

SIP BASICS

An Introduction to SIP, part 1

SIP's primary job is to control user sessions. As such, SIP contains five
primary functions that allow it to perform various session-related tasks.

The first of these functions is the user location function. UC deployments

often involve multiple networks, each containing multiple types of devices.
As such, SIP has to be able to locate the end user geographically and to
know what end systems will be used by the session.
 The second function is user availability. This function is best known for the
way that it is used in providing presence information. End users can tell the
system that they are available to talk or that they are busy and do not wish
to be disturbed.

The third function is the user capabilities function. The basic idea behind
this function is that different devices have different capabilities. For
example, there are many things that a computer is capable of doing that a
phone is not. The user capabilities function allows SIP to make a
determination of the media being used and of the parameters that are
associated with that media type. For example, will the user be
communicating using voice, video or something else?

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The fourth function is the session setup. This is the function that is
responsible for connecting a call. It establishes session parameters for both
the caller and the recipient of the call.

The fifth of the primary SIP functions is the session management function.
This is the function that allows users to end a call, transfer a call to
someone else, or make modifications to the session parameters.

Read more about SIP's five main functions.

More resources:

 SIP
DEFINITION - The Session Initiation Protocol (SIP) is an Internet
Engineering Task Force (IETF) standard protocol for initiating an
interactive user session that involves multimedia elements such as
video, voice, chat, gaming, and virtual reality. Because the SIP supports
name mapping and redirection services, it makes it possible for users to
initiate and receive communications and services from any location, and
for networks to identify the users wherever they are.

 SIMPLE
DEFINITION - SIMPLE is an add-on to SIP that stands for SIP for
Instant Messaging and Presence Leveraging Extensions. SIP was
originally developed for voice over IP (VoIP), but has since incorporated
support for Web conferencing, live video, and other media. SIMPLE is
backed by Microsoft, IBM, Sun, Novell, and other industry leaders.

 What is a user session?

DEFINITION - In tabulating statistics for Web site usage, a user session
(sometime referred to as a visit) is the presence of a user with a specific
IP address who has not visited the site recently (typically, anytime within
the past 30 minutes).

 Session border controller

DEFINITION - A session border controller (SBC) is a device or
application that governs the manner in which calls, also called sessions,
are initiated, conducted and terminated in a VoIP (Voice over Internet
Protocol) network.
 Session ID
DEFINITION - A session ID is a unique number that a Web site's server
assigns a specific user for the duration of that user's visit (session). The
session ID can be stored as a cookie, form field, or URL (Uniform
Resource Locator).

 Session initiation protocol (SIP) essentials

SIP (Session Initiation Protocol ) is the foundation of IMS (IP Multimedia
Subsystem), the architecture that could become the basis for fixed
mobile convergence. Learn the basics of SIP in this guide.

 SIP fundamentals
Session Initiation Protocol (SIP) will allow true interoperability,
eventually enabling every IP-based device and application to
communicate seamlessly with one another. This guide discusses some
of the basics of SIP, including vulnerabilities, testing and hardware.

 Why should I use SIP for VoIP?

If my application can do the work for a VoIP call established via HTTP,
why should I use SIP? SIP provides many benefits and enhanced
functionality that could be the deciding factor in making a switch -- get
acquainted with the many ways SIP could revolutionize your IP
telephony deployment.
 SIP Forum
The SIP Forum is an industry organization comprised of leaders from
leading IP communications companies that is dedicated to its mission of
advancing the adoption of products and services based on the Session
Initiation Protocol (SIP).

VOIP SIGNALING PROTOCOLS

VoIP and IP telephony signaling protocols are the codes and commands
used to establish and/or terminate calls over an IP network. These
protocols can support a variety of features including Web conferencing,
video conferencing, call waiting and transfers. These protocols are also
used to support many different multimedia applications. The most
commonly used VoIP signaling protocols are SIP, H.323 and
MGCP/MEGACO, but SIP is quickly gaining favor over its competitors.

More resources:

 Multiprotocol Label Switching (MPLS)

DEFINITION - Multiprotocol Label Switching (MPLS) is a standards-
approved technology for speeding up network traffic flow and making it
easier to manage. MPLS involves setting up a specific path for a given
sequence of packets, identified by a label put in each packet, thus
saving the time needed for a router to look up the address to the next
node to forward the packet to. MPLS is called multiprotocol because it
works with the Internet Protocol (IP), Asynchronous Transport Mode
(ATM), and frame relay network protocols.
 What is H.323?
DEFINITION - H.323 is a standard approved by the International
Telecommunication Union (ITU) in 1996 to promote compatibility in
videoconference transmissions over IP networks. H.323 was originally
promoted as a way to provide consistency in audio, video and data
packet transmissions in the event that a local area network (LAN) did
not provide guaranteed service quality (QoS).

 VoIP protocols and standards

The rapid evolution of VoIP was made possible in part by the use
of protocols and standards, or special sets of rules that end points in a
telecommunication connection use when they communicate. This guide
defines the most commonly used VoIP protocols, including signaling
protocols like SIP and H.323, that are responsible for the advancement
of IP telephony technology.

 A comparison and description of SIP and H.323

Both SIP and H.323 can be used as signaling protocols in IP networks,
but is one superior to another?

HOW DOES SIP WORK?

An Introduction to SIP, part 2

There are certain tasks that SIP is responsible for performing. SIP performs
these tasks by issuing commands, which are commonly referred to as
verbs. Any time that SIP issues a verb, the host that the verbs are being
transmitted to responds with a numerical code. This code tells SIP the
results of the requested action so that it knows what it needs to do next.
Perhaps the most commonly used verb associated with SIP is REGISTER.
REGISTER is used primarily for logging into the SIP environment, but the
REGISTER verb can also be used when a user is logging out or changing
locations.

Two other commonly used SIP verbs are SUBSCRIBE and NOTIFY. These
two verbs work together to make it possible to use presence information.

Another verb is the SERVICE verb. Any time that a user needs to change
their presence information, their client application issues a SERVICE
command to request that the host perform some kind of service. The
SERVICE command can also be used to do things like creating or
modifying conferences.

Read more about SIP verbs.

More resources:

 Application layer
DEFINITION - In the Open Systems Interconnection (OSI)
communications model, the application layer provides services for an
application program to ensure that effective communication with another
application program in a network is possible.

 SIP: Understanding the Session Initiation Protocol - Chapter 2: Introduction to

SIP
Chapter two introduces the SIP protocol. The best way to learn a
protocol is to look at examples of its use. The example message flows
included in this chapter will help you grasp some key SIP concepts.
After reading this chapter, you will better understand SIP terminology,
structures, and format.
 What are media gateways and how do H.323, SIP, MGCP and other support
protocols work?
SIP (Session Initiation Protocol) is based on RFC 2543 (Ref. 3) and is
an application layer signaling protocol. It deals with interactive
multimedia communication sessions between end users, called user
agents.

 What are the benefits of SIP trunking over T1 trunking?

SIP trunking can eliminate the need to have a traditional PSTN gateway.
But what are the benefits of SIP trunking over T1 trunking? What is
needed on a PBX to support it? Learn how SIP trunking could actually
save you money from expert Carrie Higbie.

 IP telephony: Deploying Voice over IP Protocols: Chapter 3: The Session

Initiation Protocol (SIP)
Chapter three explains the origin and purpose of the Session Initiation
Protocol (SIP). It also delves into RFC 2543 to RFC 3261 and presents
an overview of a simple SIP call, call handling services, instant
messaging, SIP security and H.323.

 Why is SIP not good for transporting large amounts of data?

As SIP becomes the protocol of choice over former favorite H.323,
concerns over its ability to transfer larger files have surfaced. Find out if
the rumors are true, and if so, how you can ensure efficient transmission
of even your largest documents.
SIP HARDWARE

SIP hardware devices include phones, IM clients and automated devices

on both client and server side. Hardware knowledge and proper selection
are important to ensure a clean VoIP or unified communications (UC)
deployment.

More resources:

 SIP phone quality and clarity solutions

Cabling and network issues can impact the quality of your VoIP service.
Can a SIP phone solve these issues? Unified communications expert
Carrie Higbie details how to troubleshoot SIP phone concerns in this
expert response.

 What is the difference between a SIP-enabled phone and an IP phone that is not
SIP-enabled?
Learn about the differences in programming a SIP-enabled phone
versus a phone that is not SIP enabled, and the costs and benefits of
each.

 VoIPuser: SIP hardware forum

This forum from VoIP User allows IT professionals to discuss SIP
hardware with their peers and receive unbiased reviews, help and
feedback.
 Hardware requirements for SIP
Learn about hardware requirements for converting a SIP call from a
provider's MPLS network to an ISDN or T1 on an inbound fax server.

SECURITY AND TROUBLESHOOTING

An Introduction to SIP, part 3

SIP can be used to establish communications between PCs, telephones
and other devices, each of which can potentially exist on separate
networks. That being the case, SIP must have some mechanism for
determining which path a packet needs to take in order to reach its
destination. SIP embeds the packet routing information into headers. There
are four primary types of headers that SIP uses: record route headers,
route headers, via headers, and contact headers.

Although a record route header is a type of routing header, it is also a type

of security mechanism. To understand why record route headers are used,
think about the role that an Office Communications Server (OCS) plays
within an organization. Oftentimes, messages between clients are routed
through an OCS 2007 server, and the server may even act as a proxy for
those messages.

Whenever a host acts as a proxy, it has the ability to place its own IP
address or fully qualified domain name into the record route header. This
tells the recipient that the host is to be used as the signaling path for all
subsequent SIP packets within that session.

This feature can act as a mechanism to help prevent session hijacking, or it

can be used for routing control. In some organizations, for example, record
route headers ensure that SIP traffic passes through a designated server
before passing through the perimeter firewall. That way, the firewall can be
configured to allow only SIP traffic to flow to and from that server. This
prevents end users from using unauthorized SIP-enabled applications such
as some instant messaging clients.

Read more about routing headers.

More resources:

 Short-circuiting hackers' SIP-based VoIP attacks

Hacker attacks against SIP-based VoIP may have been rare so far, but
as VoIP use grows, service providers need to be ready to secure their
voice networks as they route traffic without using the public switched
telephone network.

 VoIP: SIP, security and testing for your network

Ensuring that your business is getting the most from VoIP can
sometimes require a back-to-basics approach. Learn how SIP and other
tasks work to get the most from your VoIP deployment.

 VoIP protocol insecurity

SIP and other VoIP protocols like H.323 are prone to security
vulnerabilities that must be addressed to keep your system safe.

 How to use fuzzing to deter VoIP protocol attacks

Testing alone cannot defeat all attacks against VoIP. How you choose
to deploy, configure and use your VoIP products is equally important.
However, tests like these can help you reduce the inherent risk posed
by SIP and H.323 protocols.
SIP IN THE ENTERPRISE

Could a SIP system improve the way your company does business? Many
enterprises are saying that SIP has enabled a more collaborative
environment through the use of unified communications applications like
Web conferencing, allowing streamline communications processes.

More resources:

 SIP trunking
DEFINITION - Session Initiation Protocol (SIP) trunking is the use of
voice over IP (VoIP) to facilitate the connection of a private branch
exchange (PBX) to the Internet. One of the most significant advantages
of SIP trunking is its ability to combine data, voice and video in a single
line, eliminating the need for separate physical media for each mode.
The result is reduced overall cost and enhanced reliability for multimedia
services.

 SIP trunking primer

An important aspect of SIP is SIP trunking, a use of VoIP that can
minimize communication costs and maximize network usage with
unified communications. Find out why this is important for enterprises.

 We're opening a remote office -- should we get a new vendor who offers local
support and SIP instead of H.323?
SIP is becoming the most popular protocol and will provide additional
options and benefits for your roaming and remote users.
 SIP school: A to Z on SIP
Whether you're an end user or an IP network engineer, this primer will
help you clear a strategic path to SIP. You'll uncover some of the
relevant services and solutions that SIP can provide for the enterprise.
Additionally, you'll learn what IT staffs will find similar, and what they will
find different, about SIP as compared to other IP-based protocols. Since
SIP will impact many job functions within an IT organization, from the
telephony team to the security team, we've made it easy to find out what
you need to know about SIP as it relates to your responsibilities.