A SIP tool for developing SIP applications

The RADVISION SIP Family also includes:
Partial List of SIP Toolkit Features
• SIP Development Suite
A powerful and highly versatile set of tools,
• Connection reuse
• ENUM
Add-Ons and testing tools that enables developers
• Service-route/Path headers for mobile registration
to combine the necessary components for building
• Internally Multi-Threaded
an ideal development environment for an application's
• Multi-Instance
specific needs.
• TLS
• Persistent Connection
• SIP Server Platform
• Digest Authentication Support
A comprehensive SIP server development solution with
• MESSAGE Support
complete standards-based functionality of Proxy, Redirect
• UPDATE Support
and Registrar servers, as well as supporting modules like
• REFER (Transfer) Extension Support
accounting, OMAP and High Availability. SIP server platform
• High Availability (HA) Support
is perfect choice for SIP application server developers.
• SIP-T (Interworking with ISUP/QSIG) Support
• Multi-Part MIME Bodies Support
• ProLabTM Testing Suite
A versatile VoIP testing solution, based on RADVISION’'D5s
award-winning SIP Toolkit, that is suitable for use in different
stages of the product development cycle.
• PRACK (RFC 3262 - Reliable Provisional Responses)
SIP Toolkit
For developing SIP applications
• Extension Support (Manual and Automatic operation)
• 183 response
• Server-Features (Require-Supported) Mechanism
• Java SIP Toolkit
• INFO Extension Support
A powerful and highly versatile set of tools to simplify and
reduce development time of Java-based SIP applications.
• Multimedia Terminal Framework
A complete set of building blocks for developing SIP-based
IP phone applications in RTOS or embedded environments.
• RTP/RTCP Toolkit
• Multi-Homed Hosts
• IPv6 Support
• Loose Routing
• Advanced DNS Queries (Locating SIP Servers usin
SRV and NAPTR for outgoing requests* )
• SUBSCRIBE-NOTIFY (SIP Events)
• Session Timer
A standalone RTP/RTCP stack providing IPv4/IPv6, security
• Enhanced Parser
and advanced functionality.
• General URL Scheme Support (e.g.- TEL, IM)
The award-winning SIP Toolkit is a powerful and highly versatile set
of tools designed to dramatically reduce development efforts of SIP
applications. It includes all the components developers require
including a large set of quick start sample applications that demonstrate
efficient API usage, a GUI test application and detailed documentation.
The SIP Toolkit is an IETF, 3GPP and TISPAN.standards compliant,
high performance SIP implementation and provides multiple API layers
for full user control and flexibility.
The SIP Toolkit is part of RADVISION's standard-compliant SIP Developer
Suite, designed to dramatically accelerate development of SIP
applications.
• rPort
• Professional Services
A full range of design, integration and deployment
consulting services.
SIP Basics
The Session Initiation Protocol (SIP) is the industry dominant signaling
protocol for real-time communication applications such as voice over IP
(VoIP) and Instant Messaging (IM). Based on ubiquitous and accepted
Internet protocols such as SMTP and HTTP, SIP is text encoded and well
suited for the Internet and other IP environments. SIP provides the
mechanisms to implement a broad range of features including call control
services, next-generation service creation, interoperability with existing
telephony systems, and mobility.
SIP signaling functionality is divided into the following entities:
• In-Band DTMF
Products developed with the SIP Toolkit include:
• User Agents for SIP endpoint functionality
• Merging disabling
• Softswitches
• SIP Proxy for routing SIP messages to their appropriate destinations
• Transmitter object
• CSCF
• IP Address Black/White list enabled
• SIP Redirect Servers for re-directing clients to
• Gateways
• Dynamic local address setting
• MRFC
• SIP Registrar for managing user location information
• IM-MGW
• SIP Back-to-Back User Agent (B2BUA) for routing and connecting
calls with stronger control
• IP TOS setting
• A-Synchronous DNS and dynamic NDS server setting
• Dynamic Via header control
• GRUU (Globally Routable User Agent URIs) Support
• Application Servers
• BGCF
• Access Concentrators
contact an alternate set of URIs
• SIP Presence Server - Handles presence subscription requests from
watchers and notifies them about changes in presence status
• Conference Bridges
• Interactive Voice Response
• SIP-Enabled Firewall/NAT
• SIP Multimedia Servers
• 3G Cellular Phones
About RADVISION
• IP Phones
RADVISION (NASDAQ: RVSN) is the industry’s leading provider of market-proven products and technologies for unified visual communications over IP and 3G networks. With its complete
set of standards-based video networking infrastructure and developer toolkits for voice, video, data and wireless communications, RADVISION is driving the unified communications evolution
by combining the power of video, voice, data and wireless – for high definition video conferencing systems, innovative converged mobile services, and highly scalable video-enabled desktop
platforms on IP, 3G and emerging next-generation IMS networks. For more information about RADVISION, visit www.radvision.com
• 3G-SEG
• Connected PDAs
• Video Terminals
USA/Americas
T +1 201 689 6300
F +1 201 689 6301
infoUSA@radvision.com
APAC
T +852 3472 4388
F +852 2801 4071
infoAPAC@radvision.com
EMEA
T +44 (0) 20 8757 8817
F +44 (0) 20 8757 8818
infoUK@radvision.com
• Soft Phones
• Voice Enabled and e-Commerce Solutions
• Voice/Video MessagingIAD
• Session Border Controllers
Product specifications are subject to change without notice. This document is not part of a contract or license
as may be expressly agreed. RADVISION is a registered trademark of RADVISION, Ltd. All trademarks recognized.
All rights reserved. © 2007 RADVISION, Ltd. P/N 46008-00040 Rev B 07-07
Product SpecifIcations
The SIP Toolkit provides all necessary SIP, SDP and RTP services, such as
encoding, sending, parsing and receiving SIP messages over UDP, TCP
and TLS, managing SIP calls and transactions, and providing reliability.
The SIP Toolkit complies with the latest IETF and 3GPP standards.
Coded in ANSI C and cross-platform compatible, the SIP Toolkit is available
for all common operating systems.
The Toolkit features an open, object-oriented architecture, which
• Subscription Control Engine
REFER (Transfer) Extension Support
Advanced DNS Queries
makes it programmer-friendly and highly flexible. It provides
• Transaction Control Engine
REFER is a SIP method defined by RFC 3515. The REFER method
RFC 3263 (Locating SIP Servers) defines procedures for using
multiple layers of APIs including:
• Connection Management
indicates that the recipient of the REFER request should contact
advanced SRV and NAPTR DNS queries to determine the transport
• High-level APIs that hide the complexity of the protocol and
• SIP Stack Manager Layer for setting system configuration,
a third party using the contact information provided in the REFER
protocol, IP address and port at which a specific SIP server is available.
request. RFC 3515 provides a mechanism allowing the party that
These procedures can be used to dynamically update server location
is sending the REFER to be notified of the outcome of the referenced
and to implement redundancy among servers for fault tolerance or
load balancing.
enable rapid development of applications.
memory allocation, logging and other resources.
• Mid- and low-level APIs that expose the internals of the
protocol and allow for more power and customization based
Enhanced Features
request with a NOTIFY request. This implementation uses subscription
on application requirements.
TLS
objects for REFER implementation.
TLS is a security protocol which is typically layered on top of
SIP Toolkit Architecture
connection-oriented transports such as TCP. TLS allows client/server
applications to communicate over TCP in a way that is designed to
SIP Stack
prevent eavesdropping, tampering, or message forgery. TLS provides
The SIP Stack is an internally multi-threaded (configurable)
a solution for many of the security issues SIP applications face and
library containing all SIP-specific functionality of the Toolkit
ties in well with the existing SSL/TLS infrastructure that serves HTTP
including message encoding and decoding, transaction and
applications. SIP uses SIPS: URL scheme for TLS addresses.
SUBSCRIBE-NOTIFY (SIP Events)
Connection reuse
RFC 3265 (SIP Specific Event Notification) is a SIP extension that
Connection reuse is a networking feature that results double network
allows for subscription and event notifications using SIP. SIP Events
efficiency. The working principle is using one network connection
an important infrastructure for services such as Presence and
(TCP socket) for SIP requests in two directions.
Message Waiting Indication.
High Availability (HA) Support
Enhanced Parser Functionality
The SIP Toolkit provides the necessary building blocks for implementing
Enables the application to accept or fix bad message syntax. This
Persistent Connection
Highly Available systems that can recover from machine failure
provides ability to handle corrupted message parts and interwork
SDP (Session Description Protocol) Stack
In many cases, a single TCP connection may be reused for different
without losing call state. The SIP stack enables creation of one or
with proprietary implementations.
The SDP Stack is a library for SDP message processing. The SDP Stack
messages/transactions/dialogs. Opening and closing TCP connections
was written in compliance with RFC 2327 and it enables
more replicated/ redundant network entities (for standby or parallel
often is not desirable because of the extra messaging overhead of
parsing/encoding of any SDP message field.
the TCP handshake (and even more so in TLS connections).
call management and SIP extensions.
ENUM
When SIP users dial a phone number, the SIP network has to convert
it to SIP addresses for routing purpose. This advanced feature is done
automatically by SIP Toolkit as defined in RFC 3764.
operation) and seamless switching between them.
Standards Supported (partial list)
• IETF RFC 3261 (SIP: Session Initiation Protocol)
SIP-T (Interworking with ISUP/QSIG) Support
• IETF RFC 3262 (Reliability of Provisional Responses in
SIP-T is an IETF umbrella specification that utilizes different SIP
• Session Initiation Protocol (SIP))
extensions and advanced capabilities (such as PRACK, 183, INFO,
• IETF RFC 3263 (Locating SIP Servers)
Multi-Part MIME, Server Features) in order to interwork SIP with
• IETF RFC 3264 (An Offer/Answer Model with Session
SS7/ISUP or QSIG networks. The SIP Toolkit provides the entire feature
Description Protocol (SDP))
set and simplifies enabling SIP-T functionality in any application
• IETF RFC 3265 (SIP Specific Event Notification)
rapidly.
• ETF RFC 3266 (Support for IPv6 in Session Description
Lean footprint
Latest SIP Toolkit release introduces reduced memory footprint
achieved several techniques like:
Optimized SIP parsing engine, SDP optimization and other techniques.
Protocol (SDP))
• IETF RFC 2327 (SDP–'D0 Session Description Protocol)
• IETF RFC 1889 and 1890 (RTP/RTCP)
• Numerous Internet Drafts for Various SIP Extensions
• Dozens of IETF and 3GPP SIP and SDP extensions
PRACK (RFC 3262- Reliable Provisional Responses)
Extension Support
The SDP Stack also provides an SDP Message Layer for
PRACK (PRovisional ACKnowledgment) is an IETF SIP extension for
creating, browsing and editing SDP message parts.
sending provisional responses reliably. PRACK is useful opening one
way media sessions before call establishment and QoS negotiation
RTP/RTCP Stack*
before completing the INVITE transaction.
The RTP/RTCP Stack is a library for sending and receiving
RTP and RTCP packets.
*RADVISION also offers a standalone Advanced RTP/RTCP
General URL Scheme Support
(RFC3550/3551 compliant) Toolkit providing IPv6 and other
SIP defines and uses different URL schemes such as SIP, IM, TEL. In
advanced functionality like secured RTP (SRTP as defined in
addition, some implementations define proprietary URL schemes. A
RFC 3711).
general framework in the SIP Stack provides support for sending and
IPv6 Support
The SIP Stack fully supports both IPv4 and IPv6 as underlying
protocols and can be used seamlessly with either type of network
simultaneously.
receiving any type of URL scheme. In Particular, SIP, SIPS, TEL, PRES
SIP Toolkit APIs
The SIP Toolkit is standards based and enhanced with intuitive object
and IM URIs are implement and all the other URI schemes are
supported.
The SIP Toolkit is delivered with:
Operating Systems (partial list)
• Source Code
• Windows 2000/2003/XP
• Nucleus
data has to be exchanged. This is done by key management framework.
• Sample Programs
• Windows Mobile
• pSOS
In particular MIKEY (RFC 3830) protocol is implemented for SRTP
• GUI Test Application (soft phone UA) with full signaling capabilities
• Windows Vista
support.
• Complete Documentation
• Linux Redhat / SUSE
• Embedded Linux (Monta
Vista)
• VxWorks
• 64 bit operating systems
oriented APIs to provide optimal control over SIP Stack activities.
SDP key management extension
Highlights of the Toolkit’'D5s APIs include:
Before the initiation of secured media streams (SRTP), cryptographic
• SIP Message Layer for creating, browsing, editing and
comparing SIP messages and message parts
• Dialog Control Engine for rapid SIP application development
• Solaris
The Toolkit features an open, object-oriented architecture, which
• Subscription Control Engine
REFER (Transfer) Extension Support
Advanced DNS Queries
makes it programmer-friendly and highly flexible. It provides
• Transaction Control Engine
REFER is a SIP method defined by RFC 3515. The REFER method
RFC 3263 (Locating SIP Servers) defines procedures for using
multiple layers of APIs including:
• Connection Management
indicates that the recipient of the REFER request should contact
advanced SRV and NAPTR DNS queries to determine the transport
• High-level APIs that hide the complexity of the protocol and
• SIP Stack Manager Layer for setting system configuration,
a third party using the contact information provided in the REFER
protocol, IP address and port at which a specific SIP server is available.
request. RFC 3515 provides a mechanism allowing the party that
These procedures can be used to dynamically update server location
is sending the REFER to be notified of the outcome of the referenced
and to implement redundancy among servers for fault tolerance or
load balancing.
enable rapid development of applications.
memory allocation, logging and other resources.
• Mid- and low-level APIs that expose the internals of the
protocol and allow for more power and customization based
Enhanced Features
request with a NOTIFY request. This implementation uses subscription
on application requirements.
TLS
objects for REFER implementation.
TLS is a security protocol which is typically layered on top of
SIP Toolkit Architecture
connection-oriented transports such as TCP. TLS allows client/server
applications to communicate over TCP in a way that is designed to
SIP Stack
prevent eavesdropping, tampering, or message forgery. TLS provides
The SIP Stack is an internally multi-threaded (configurable)
a solution for many of the security issues SIP applications face and
library containing all SIP-specific functionality of the Toolkit
ties in well with the existing SSL/TLS infrastructure that serves HTTP
including message encoding and decoding, transaction and
applications. SIP uses SIPS: URL scheme for TLS addresses.
SUBSCRIBE-NOTIFY (SIP Events)
Connection reuse
RFC 3265 (SIP Specific Event Notification) is a SIP extension that
Connection reuse is a networking feature that results double network
allows for subscription and event notifications using SIP. SIP Events
efficiency. The working principle is using one network connection
an important infrastructure for services such as Presence and
(TCP socket) for SIP requests in two directions.
Message Waiting Indication.
High Availability (HA) Support
Enhanced Parser Functionality
The SIP Toolkit provides the necessary building blocks for implementing
Enables the application to accept or fix bad message syntax. This
Persistent Connection
Highly Available systems that can recover from machine failure
provides ability to handle corrupted message parts and interwork
SDP (Session Description Protocol) Stack
In many cases, a single TCP connection may be reused for different
without losing call state. The SIP stack enables creation of one or
with proprietary implementations.
The SDP Stack is a library for SDP message processing. The SDP Stack
messages/transactions/dialogs. Opening and closing TCP connections
was written in compliance with RFC 2327 and it enables
more replicated/ redundant network entities (for standby or parallel
often is not desirable because of the extra messaging overhead of
parsing/encoding of any SDP message field.
the TCP handshake (and even more so in TLS connections).
call management and SIP extensions.
ENUM
When SIP users dial a phone number, the SIP network has to convert
it to SIP addresses for routing purpose. This advanced feature is done
automatically by SIP Toolkit as defined in RFC 3764.
operation) and seamless switching between them.
Standards Supported (partial list)
• IETF RFC 3261 (SIP: Session Initiation Protocol)
SIP-T (Interworking with ISUP/QSIG) Support
• IETF RFC 3262 (Reliability of Provisional Responses in
SIP-T is an IETF umbrella specification that utilizes different SIP
• Session Initiation Protocol (SIP))
extensions and advanced capabilities (such as PRACK, 183, INFO,
• IETF RFC 3263 (Locating SIP Servers)
Multi-Part MIME, Server Features) in order to interwork SIP with
• IETF RFC 3264 (An Offer/Answer Model with Session
SS7/ISUP or QSIG networks. The SIP Toolkit provides the entire feature
Description Protocol (SDP))
set and simplifies enabling SIP-T functionality in any application
• IETF RFC 3265 (SIP Specific Event Notification)
rapidly.
• ETF RFC 3266 (Support for IPv6 in Session Description
Lean footprint
Latest SIP Toolkit release introduces reduced memory footprint
achieved several techniques like:
Optimized SIP parsing engine, SDP optimization and other techniques.
Protocol (SDP))
• IETF RFC 2327 (SDP–'D0 Session Description Protocol)
• IETF RFC 1889 and 1890 (RTP/RTCP)
• Numerous Internet Drafts for Various SIP Extensions
• Dozens of IETF and 3GPP SIP and SDP extensions
PRACK (RFC 3262- Reliable Provisional Responses)
Extension Support
The SDP Stack also provides an SDP Message Layer for
PRACK (PRovisional ACKnowledgment) is an IETF SIP extension for
creating, browsing and editing SDP message parts.
sending provisional responses reliably. PRACK is useful opening one
way media sessions before call establishment and QoS negotiation
RTP/RTCP Stack*
before completing the INVITE transaction.
The RTP/RTCP Stack is a library for sending and receiving
RTP and RTCP packets.
*RADVISION also offers a standalone Advanced RTP/RTCP
General URL Scheme Support
(RFC3550/3551 compliant) Toolkit providing IPv6 and other
SIP defines and uses different URL schemes such as SIP, IM, TEL. In
advanced functionality like secured RTP (SRTP as defined in
addition, some implementations define proprietary URL schemes. A
RFC 3711).
general framework in the SIP Stack provides support for sending and
IPv6 Support
The SIP Stack fully supports both IPv4 and IPv6 as underlying
protocols and can be used seamlessly with either type of network
simultaneously.
receiving any type of URL scheme. In Particular, SIP, SIPS, TEL, PRES
SIP Toolkit APIs
The SIP Toolkit is standards based and enhanced with intuitive object
and IM URIs are implement and all the other URI schemes are
supported.
The SIP Toolkit is delivered with:
Operating Systems (partial list)
• Source Code
• Windows 2000/2003/XP
• Nucleus
data has to be exchanged. This is done by key management framework.
• Sample Programs
• Windows Mobile
• pSOS
In particular MIKEY (RFC 3830) protocol is implemented for SRTP
• GUI Test Application (soft phone UA) with full signaling capabilities
• Windows Vista
support.
• Complete Documentation
• Linux Redhat / SUSE
• Embedded Linux (Monta
Vista)
• VxWorks
• 64 bit operating systems
oriented APIs to provide optimal control over SIP Stack activities.
SDP key management extension
Highlights of the Toolkit’'D5s APIs include:
Before the initiation of secured media streams (SRTP), cryptographic
• SIP Message Layer for creating, browsing, editing and
comparing SIP messages and message parts
• Dialog Control Engine for rapid SIP application development
• Solaris
The RADVISION SIP Family also includes:
Partial List of SIP Toolkit Features
• SIP Development Suite
A powerful and highly versatile set of tools,
• Connection reuse
• ENUM
Add-Ons and testing tools that enables developers
• Service-route/Path headers for mobile registration
to combine the necessary components for building
• Internally Multi-Threaded
an ideal development environment for an application's
• Multi-Instance
specific needs.
• TLS
• Persistent Connection
• SIP Server Platform
• Digest Authentication Support
A comprehensive SIP server development solution with
• MESSAGE Support
complete standards-based functionality of Proxy, Redirect
• UPDATE Support
and Registrar servers, as well as supporting modules like
• REFER (Transfer) Extension Support
accounting, OMAP and High Availability. SIP server platform
• High Availability (HA) Support
is perfect choice for SIP application server developers.
• SIP-T (Interworking with ISUP/QSIG) Support
• Multi-Part MIME Bodies Support
• ProLabTM Testing Suite
A versatile VoIP testing solution, based on RADVISION’'D5s
award-winning SIP Toolkit, that is suitable for use in different
stages of the product development cycle.
• PRACK (RFC 3262 - Reliable Provisional Responses)
SIP Toolkit
For developing SIP applications
• Extension Support (Manual and Automatic operation)
• 183 response
• Server-Features (Require-Supported) Mechanism
• Java SIP Toolkit
• INFO Extension Support
A powerful and highly versatile set of tools to simplify and
reduce development time of Java-based SIP applications.
• Multimedia Terminal Framework
A complete set of building blocks for developing SIP-based
IP phone applications in RTOS or embedded environments.
• RTP/RTCP Toolkit
• Multi-Homed Hosts
• IPv6 Support
• Loose Routing
• Advanced DNS Queries (Locating SIP Servers usin
SRV and NAPTR for outgoing requests* )
• SUBSCRIBE-NOTIFY (SIP Events)
• Session Timer
A standalone RTP/RTCP stack providing IPv4/IPv6, security
• Enhanced Parser
and advanced functionality.
• General URL Scheme Support (e.g.- TEL, IM)
The award-winning SIP Toolkit is a powerful and highly versatile set
of tools designed to dramatically reduce development efforts of SIP
applications. It includes all the components developers require
including a large set of quick start sample applications that demonstrate
efficient API usage, a GUI test application and detailed documentation.
The SIP Toolkit is an IETF, 3GPP and TISPAN.standards compliant,
high performance SIP implementation and provides multiple API layers
for full user control and flexibility.
The SIP Toolkit is part of RADVISION's standard-compliant SIP Developer
Suite, designed to dramatically accelerate development of SIP
applications.
• rPort
• Professional Services
A full range of design, integration and deployment
consulting services.
SIP Basics
The Session Initiation Protocol (SIP) is the industry dominant signaling
protocol for real-time communication applications such as voice over IP
(VoIP) and Instant Messaging (IM). Based on ubiquitous and accepted
Internet protocols such as SMTP and HTTP, SIP is text encoded and well
suited for the Internet and other IP environments. SIP provides the
mechanisms to implement a broad range of features including call control
services, next-generation service creation, interoperability with existing
telephony systems, and mobility.
SIP signaling functionality is divided into the following entities:
• In-Band DTMF
Products developed with the SIP Toolkit include:
• User Agents for SIP endpoint functionality
• Merging disabling
• Softswitches
• SIP Proxy for routing SIP messages to their appropriate destinations
• Transmitter object
• CSCF
• IP Address Black/White list enabled
• SIP Redirect Servers for re-directing clients to
• Gateways
• Dynamic local address setting
• MRFC
• SIP Registrar for managing user location information
• IM-MGW
• SIP Back-to-Back User Agent (B2BUA) for routing and connecting
calls with stronger control
• IP TOS setting
• A-Synchronous DNS and dynamic NDS server setting
• Dynamic Via header control
• GRUU (Globally Routable User Agent URIs) Support
• Application Servers
• BGCF
• Access Concentrators
contact an alternate set of URIs
• SIP Presence Server - Handles presence subscription requests from
watchers and notifies them about changes in presence status
• Conference Bridges
• Interactive Voice Response
• SIP-Enabled Firewall/NAT
• SIP Multimedia Servers
• 3G Cellular Phones
About RADVISION
• IP Phones
RADVISION (NASDAQ: RVSN) is the industry’s leading provider of market-proven products and technologies for unified visual communications over IP and 3G networks. With its complete
set of standards-based video networking infrastructure and developer toolkits for voice, video, data and wireless communications, RADVISION is driving the unified communications evolution
by combining the power of video, voice, data and wireless – for high definition video conferencing systems, innovative converged mobile services, and highly scalable video-enabled desktop
platforms on IP, 3G and emerging next-generation IMS networks. For more information about RADVISION, visit www.radvision.com
• 3G-SEG
• Connected PDAs
• Video Terminals
USA/Americas
T +1 201 689 6300
F +1 201 689 6301
infoUSA@radvision.com
APAC
T +852 3472 4388
F +852 2801 4071
infoAPAC@radvision.com
EMEA
T +44 (0) 20 8757 8817
F +44 (0) 20 8757 8818
infoUK@radvision.com
• Soft Phones
• Voice Enabled and e-Commerce Solutions
• Voice/Video MessagingIAD
• Session Border Controllers
Product specifications are subject to change without notice. This document is not part of a contract or license
as may be expressly agreed. RADVISION is a registered trademark of RADVISION, Ltd. All trademarks recognized.
All rights reserved. © 2007 RADVISION, Ltd. P/N 46008-00040 Rev B 07-07
Product SpecifIcations
The SIP Toolkit provides all necessary SIP, SDP and RTP services, such as
encoding, sending, parsing and receiving SIP messages over UDP, TCP
and TLS, managing SIP calls and transactions, and providing reliability.
The SIP Toolkit complies with the latest IETF and 3GPP standards.
Coded in ANSI C and cross-platform compatible, the SIP Toolkit is available
for all common operating systems.
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