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
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?
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.
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.
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.
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.
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.
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.
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 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.
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.
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.
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.
Dig deeper on SIP and Unified Communications Standards