The Radisys Application Server represents an entirely new approach to deploying enhanced services in IMS–enabled, converged TDM/IP and end–to–end VoIP carrier networks. Its open architecture and innovative use of next–generation network standards—including SIP, IMS, VoiceXML, Diameter and XML—provide a multi–service platform with tremendous flexibility and scalability.
Radisys provides a complete solution for deploying revenue–generating IP voice communication services in converged TDM/IP and end-to-end IP packet networks. Based on our open, extensible and highly scalable Radisys Application Server and Radisys Media Server, the Radisys hosted communications solution includes an expanding suite of turnkey SIPware services—or create your own innovative services using the Radisys Service Creation Environment.
Instead of costly circuit–switched, single–application platforms, carriers can deliver multiple SIP–based services starting with a single Application Server – and then rapidly add or modify services as needed, with no service interruption.
The Radisys approach provides a flexible, pure–IP solution for replacing legacy trunk–side services traditionally accessed through toll–free numbers or enhanced services in wireless networks using IMS interfaces. It also enables emerging carriers implementing a full range of competitive services to be competitive from day one, whether acquiring wireless TDM–based or VoIP subscribers.
A Complete IP Voice Service Solution
Radisys provides a complete solution for deploying revenue–generating IP voice services in converged TDM/IP and end–to–end VoIP networks. Based on our open, extensible and highly scalable Service Delivery Platform and Media Server, the Radisys solution includes an expanding suite of turnkey SIPware services—or create your own innovative services using the Radisys Service Creation Environment.
High Capacity, Carrier-Ready
The Radisys Application Server was designed from the ground up for high volume, high-performance carrier networks. It can run up to over 7,000 concurrent call sessions on a single Linux system, or scale to extremely large call handling capacities by configuring multiple servers. Robust features include a high-performance service logic execution engine (XTML Server) written in C++, a sophisticated thread-pooling scheme for executing call sessions, compiled-to-memory service logic scripts represented as XTML and choice of Linux -based runtime environment.
Flexible IP Integration
The Application Server is based on a pure-IP design with integrated Session Initiation Protocol (SIP) and Media Gateway Control Protocol (MGCP) support. Radisys developed, high-performance SIP and MGCP protocol stacks written in C++ allow the Application Server to interact with SIP-enabled media gateways, IP phones and SIP user agents and control IP media servers using SIP/VoiceXML or SIP/MSML protocols.
Distributed Call Processing and Load Balancing
Pure-IP design allows distribution of call processing across many Application Servers that can be configured in a high-availability cluster, co-located or distributed across an IP network. For load balancing, an integrated SIP proxy server directs calls to the least busy Application Server in the cluster. This enables a single service access point comprised of a cluster of Application Servers to scale up to well over 50,000 simultaneous calls-enough to process 1 billion minutes in a single node!
High Availability without High Third-party Costs
Two levels of high availability are offered when configuring Application Servers to ensure SIPware (or third-party) services stay running under any failure scenario. The first level configures Application Servers in an N+1 configuration within a cluster, insuring that any one server failure will not impact overall system availability.
The second level, CallComplete, maintains all calls in progress if any Application Server fails using 1:1 server redundancy by saving call session state information on a hot standby. For service providers, third-party developers and OEMs looking to differentiate their service offerings, the Application Server offers an exceptionally flexible, scalable and cost-effective solution for deploying unique, revenue-generating subscriber voice services.
Extensible, Plug-in Architecture
Radisys provides a Software Development Kit (SDK) that defines a lightweight, component-based framework for “plugging in” new software modules to enable new application features. Using the SDK, developers can add components to both the Service Creation Environment (SCE) and the Application Server, extending the SCE development framework to generate the XTML that supports new features added to the Application Server.
Centralized System Management
The System Management Console enables service providers to monitor and manage their Application Servers and their associated media servers via a standard Web browser. Several Application Servers can be monitored from a single view, and XTML servers and applications can be started and stopped. Applications can be brought out of service gracefully, servicing all existing requests before shutting down, or they can be stopped immediately. The number of call sessions executing for each application can be modified while the application is running.
SIP Compliance
- IETF Requests for Comments RFC (RFC) compliance
- RFC 3261 – Session Initiation Protocol (SIP)
- RFC 3264 – An Offer/Answer Model with the Session Description Protocol (SDP)
- RFC 2327 – Session Description Protocol (SDP)
- RFC 3262 – Reliability of Provisional Responses in SIP
- RFC 2976 – SIP INFO Method
- RFC 3265 – SIP-specific Event Notification (not fully compliant)
- RFC 3515 – SIP Refer method
- RFC 3892 – SIP Referred-by Mechanism
- RFC 3891 – SIP “Replaces” Header
- RFC 3842 – Message Summary and Message Waiting Indication Event Package for SIP
- RFC 2806 – Tel Uniform Resource Identifiers (URI): for Telephony Calls
- RFC 3389 – Real-time Transport Protocol (RTP) Payload for Comfort Noise (CN)
- RFC 3665 – SIP Basic Call Flow Examples
- RFC 2045/46 – Multipurpose Internet Mail Extensions (MIME) Parts One and Two
- RFC 2396 – Uniform Resource Identifiers (URI): Generic Syntax
- RFC 1889 – RTP: Transport Protocol for Real-Time Applications
- RFC 3666 – SIP PSTN Call Flows
- RFC 2617 – HTTP Authentication: Basic and Digest Access Authentication
- RFC 2833 – RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
- RFC 3550 – RTP: Transport Protocol for Real-Time Applications
- RFC 3489 – STUN – Simple Traversal of UDP Through Network Address Translators RFC
- 3323 – Privacy Mechanism for SIP
- RFC 3863 – Presence Information Data Format (PIDF)
IMS-specific SIP Standards Compliance
- RFC 3325 – Private Extensions to SIP for Asserted Identity within Trusted Networks
- RFC 3311 – SIP UPDATE Method
- RFC 3312 – Integration of Resource Management and Session Invitation Protocol
- RFC 3891 – SIP “Replaces” Header
- RFC 4032 – Update to SIP Preconditions Framework
- RFC 3986 – Uniform Resource Identifier (URI): Generic Syntax
- RFC 4004/05/06 – Diameter: Proper handling of vendor-specific applications and AVPs
3GPP Technical Specs
- IMS Sh interface for subscriber information.
- 3GPP TS 29.329 (Sh interface)
- IMS rf interface for offline charging.
- 3GPP TS 32.260 (Ro/Rf charging)*
- IMS ISC interface for session control.
- 3GPP TS 24.228 (core signaling)
- 3GPP TS 24.229 (IMS call control)
Application Server Specifications
- Red Hat Linux Configuration
- IBM X Series or Dell PowerEdge Servers
- Number of sessions/simultaneous calls: up to 7,000 depending on application
- SIP requests per second: 450
