The Aricent SIP Server Framework (SIP-SF) is an
award-winning framework that speeds development of reliable, scalable and
feature-rich SIP applications. The Aricent SIP-SF comprises two components:
- SIP Core Infrastructure Server
- SIP Application Server
SIP-SF serves as a base for proxy as well as any other
SIP-based server entity, thereby reducing time-to-market by providing a
hassle-free infrastructure. Aricent SIP-SF:
- Implements all the core SIP functionality
including state information automatically.
- Provides a carrier class framework that easily
allows new components to be added in and out dynamically as well as
statically.
The SIP-SF is a powerful yet flexible solution. It
provides tested and standards-compliant core SIP functionalities such as
Call State Control Function (CSCF), Proxy, Registrar, Redirect, B2BUA, and
Location Server. More importantly, SIP-SF allows application developers to
build upon these core functionalities and deliver niche applications.
Aricent’s Companion Lifecycle Services
The majority of Aricent’s products are delivered with
tailored combinations of our Lifecycle Services, including Global Innovation
and Design, Product Development, Testing and Certification, Network
Engineering, Maintenance and Support, and Business Operations and Systems
Integration. Aricent’s engineers and consultants have successfully completed
thousands of services engagements globally using flexible delivery models
ranging from on-site to off-shore. Aricent’s services offer deep
communications domain expertise, cost and time efficiencies, quick ramp up
and contemporary commercial engagement models including end-to-end
commitments.
The SIP-SF solution isolates the application developer
from all SIP-specific protocol and behavioral details, enabling the
developer to focus on the service to be provided. Aricent's carrier-class,
proven solution thus helps customers reduce service development time and
costs.
One of the most proven and mature products in the
industry, SIP-SF handles live traffic for the largest service provider’s
network and for one of the largest Internet Telephony Service Provider
(ITSP) networks in Japan.
Aricent has also introduced Centrex functionalities in
its SIP-SF. The Centrex feature enables equipment manufacturers to build a
hosted PBX solution in minimal time. The Aricent SIP Server Framework with
Centrex functionality is a powerful development accelerator, with a
sophisticated dial plan, a hunt group, privileged profiling, and
intranet/extranet calling.
SIP-SF is an ideal platform for hosting SIP services.
SIP-SF is a ready-to-deploy SIP Proxy, Registrar, Redirect, Presence and
Location Server. Aricent's SIP-SF today provides several advanced features,
including SIP Centrex (Hosted PBX) and Per-User Call Processing Language,
all of which render tremendous value in using it as an underlying platform
to build new services while significantly reducing time, risk, and cost to
market.
Out of the box, the SIP-SF meets virtual IP redundancy,
SIP load balancing, SNMP manageability, congestion control, emergency
calling, and other real-life network requirements. Targeted at OEMs, SIP-SF
also delivers the flexibility and extensibility that OEMs need to deliver
differentiated products to the market.
Aricent's SIP server is a dual play architecture,
positioned as both Core Infrastructure and Application Development
Framework.
Aricent’s SIP Server can be positioned in two ways:
- As the SIP core infrastructure element to build
basic infrastructure needed in a SIP Network or using its B2BUA feature
set
- As an application infrastructure to build
applications, such as collaboration, CTI, and gaming.
The SIP-SF thus enables easy integration of multiple
call models, whether BCSM-based or RFC 3261-based. Service providers can use
any call model, depending on the kind of service being executed. Further,
the SIP-SF can be made to interact with an SCN using a regular BCSM-based
model and at the same time, perform forking functionality of proxy using the
regular 3261 UAC/UAS model.
The key architectural features of the SIP Server
Framework are:
- High scalability – its performance increases with
an increase in the number of CPUs
- Can support multiple CSMs (service logic)
simultaneously
- Support for user-defined service logic
- Component-based architecture- can chain components
as desired for future enhancements
- Supports distributed operation across various
nodes
- Centralized configuration of all components
- Message-based interfaces
- Threaded model-thread pool management
- Easy integration with third party modules- all
functional modules implemented over APIs
- Compliant to ISO C++ and EC++ standards to ensure
maximum portability
- Ready-to-deploy SIP Proxy, Registrar, and Redirect
server
- Value-added features including SIP Centrex, Call
Processing Language, and Presence server
- Integrated load balancing and high availability
platform
- Multi operating system portability
- Easy management - SNMP, API, XML
- Multiple database support - ODBC
- Supports both IPv4 and IPv6.
- TLS security and digest authentication
- Support for emergency calls and priority routing
- Implemented in C++ to offer complete
object-oriented interface
- Configuration through Number Translation Markup
Language (NTML)
- 'SIP: Session Initiation Protocol', RFC 3261
- 'HTTP Authentication: Basic and Digest Access
Authentication', RFC 2617
- 'Session Initiation Protocol (SIP): Locating SIP
Servers', RFC 3263
- 'CPL: A Language for User Control of Internet
Telephony Services', draft-ietf-iptel-cpl-06
- 'URLs for Telephone Calls', RFC 2806
- 'Management Information Base for Session
Initiation Protocol', draft-ietf-sip-mib-04.txt
- Source Code for the SIP Core Framework Platform
- OS Wrappers for Sun Solaris or Linux
- Administration guide
- Installation manual
- Interface control documents
- Release notes
- Optional: Add-on feature modules such as Centrex,
CPL, Application Server, etc.
- Standard and customized support plans
- Training and on-site consultancy
- Turnkey integration and porting services