Session Initiation Protocol - SIP
Session initiation protocol is a
set of text commands and processes that work with applications (application layer) to setup, manage, and terminate communication sessions. SIP is a simplified version of the ITU H.323 packet multimedia system. SIP is defined in RFC 2543.
This figure shows how a SIP system uses relatively simple
text messages to setup and control telephone calls. This diagram shows how a telephone has SIP capability that is controlled by a call server. This SIP based telephone is called a User Agent (UA). The User Agent (UA) is actually a gateway that converts audio (e.g. sound) and control information (e.g. dialed digits) into packets that can be routed through a data network (such as the Internet) to call servers and other User Agents (UAs.) The control packets are sent to and from the call server to request and receive calls. Call servers may communicate with other call servers to setup distant call connections. This diagram shows how a distant call server controls a User Agent (UA) gateway that allows calls to connect from the Internet to another telephone.
Session Initiation
Protocol - SIP System Diagram
SIP and its associated protocols were
designed to allow for changes that will permit new communication services
and enhanced systems integration. To enhance the capabilities of SIP
systems, the basic SIP protocol can be extended through the use of
protocol extensions and SIP development toolkits. Some of the key
extensions for SIP include SIP for Instant Messaging and Presence
Leveraging Extensions (SIMPLE).
The core SIP protocol was intentionally
designed to be simple in its design and to allow for future extensibility.
It is through this extensibility that many of the new SIP services and
features are appearing.
Changes to SIP and supporting protocols
are made by the Internet Engineering Task Force (IETF) SIP working group.
Because the addition of new features and capabilities to SIP can
potentially cause unexpected equipment operation results, the SIP working
group extends the protocol when there is significant documented reasons to
extend the protocol.
SIP Extensions
SIP Extensions are a set of commands or
protocols that are used to extend the capabilities of the core SIP
protocol. SIP extensions are commonly used to rapidly extend the
capabilities of an existing application or protocol without changing the
underlying application or protocol.
This figure shows how SIP extensions can
be used to enhance the capabilities of the SIP core protocol and how they
interact with the existing core protocol. This diagram shows that SIP
extensions are used to process new types of messages that are sent on the
SIP system. When core messages are sent between the SIP devices (such as
between proxy servers), they are processed for the function defined in the
standard. If a new type of message is received, the SIP device will
determine if the message is associated with a SIP extension that is has
installed. This diagram shows that the message processed by the SIP
extension may interact with the core protocol and other SIP extensions.
SIP Extensions Diagram
Related Session Initiation
Protocol - SIP Terms
Access Control List - ACL
Access Gateway - AGW
Access Router Card - ARC
Adaptive Jitter Buffer
Analog Telephone Adapter - ATA
Application Server - AS
Automatic Discovery - Auto Discovery
Call Agent
Call Servers
Central Office - CO
Click to Dial or Click to Call
Coder/Decoder - Codec
Communication Server
Communications Assistance For Law Enforcement - CALEA
Competitive Local Exchange Carrier - CLEC
Conference Bridge
Connected State
Country Code
Dial Map
Dial Plan
Dialed Number Identification Service - DNIS
Direct Inward Dialing - DID
Direct Station Select - DSS
Dynamic Call Routing
Dynamic Host Configuration Protocol - DHCP
Dynamic IP Addressing
Dynamic Traffic Routing
End User Devices - User
Agents
Equipment Configuration
Ethernet IP Phone
Find Me Service
Foreign Exchange Office - FXO
Foreign Exchange Station - FXS
Forking Proxy Server
Forwarding Delay
Fragmentation
G.711
G.723.1
G.729
Gateway - GW
General Switched Telephone Network - GSTN
Hosted Telephony
Interexchange Carrier - IXC
International Toll Free Service - ITFS
Internet Phone - IPhone
Internet Protocol Detail Record - IPDR
Internet Protocol Phone - IP Phone
Internet Protocol Private Branch Exchange - IPBX or IP PBX
Internet Telephony Service Provider - ITSP
IP Enabled PBX
Key Service Unit - KSU
Location Servers - LS
Mailbox - MBX
Message Waiting Indicator - MWI
Network Address Translation - NAT
Network Operations Center - NOC
Policy Server
Presence Server
Presence User Agent - PUA
Presentity - Presence Entity
Protocol Extension
Proxy Servers
Quality Of Service - QoS
Real Time Transport Protocol - RTP
Redirection Servers
Registrars
Residential Gateway - RG
Route Update Protocol - RU
Scheduling Algorithm
Secure Server
Secure Sockets Layer - SSL
Sequenced Packet Protocol - SPP
Service Activation
Service Level Agreement - SLA
Service Provider Portability
Service Requirements
Services URL
Servlet
Session
Session Border Controller - SBC
Session Configuration Protocol - SCP
Session Description Protocol Next Generation - SDPng
Session Initiation Protocol - SIP
Session Layer
Session Management Protocol - SMP
Signaling Network
Signaling Protocol
SIP for Instant Messaging and Presence Leveraging Extensions - SIMPLE
SIP Info
SIP-T
SNMP Trap
Soft Switch - Softswitch
Softphone
Static IP Addressing
Stream Protocol
Streaming
Streaming Media
Streaming Protocol
Streaming Services
Subnet
Switching
T1 Carrier - T-1
Telephony
Session Initiation Protocol -
SIP Books
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more
details
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Introduction
to SIP IP Telephony Systems)
This
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function parts of SIP systems and operations.
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