Trillium SIP - IMS Protocol & IMS Protocol Software

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The Session Initiation Protocol (SIP) (RFC3261) defined by IETF is an application layer control protocol that can establish, modify and terminate multimedia sessions or calls. SIP aims to reduce signaling complexity while delivering internet-like services to telecommunications users. SIP has been adopted by all major standards bodies as the core signaling protocol for Next Generation Networks predominately based on IMS protocol specifications, including, but not limited to:

• 3GPP & 3GPP2
• ITU-T
• ETSI

SIP is a text-based client-server protocol modeled after the Hypertext Transfer Protocol (http). SIP can invite parties to both unicast and multicast sessions and is independent of the type of session being established. It supports next-generation services like multimedia distribution (i.e. voice, video, audio, and messaging), multimedia conferencing, and presence. Trillium SIP is the key product within the Trillium IMS protocol software family and supports all the features and extensions required by IMS protocol specifications.

Trillium SIP software is a C source code implementation of the SIP protocol stack for use in next-generation telecommunications platforms. The protocol software includes:

• User Agent (UA) and Network Server (NS) configurations
  - User Agent, B2BUA, Call Stateful Proxy, Transaction Stateful Proxy, Stateless Proxy, Redirect, and Registrar Server
• SIP protocol software layer
  - Encode/decode, session state management, SIP protocol features, and over 40 SIP extensions
  - Provides support for all SIP IMS protocol software requirements
• Trillium XML based test architecture (XTA) for simplified acceptance and regression protocol testing
  - Trillium XTA provides sample configuration files for integrated SIP over UDP/TCP and SIP over SCTP that speed the integration and development process.
  - Configuration files can be used as-is or customized by customers.
• Six (6) months warranty and support
• Documentation and training

Trillium SIP software enables telecom equipment manufacturers of VoIP and IMS network equipment to:

• Accelerate time to market
• Reduce initial and incremental development and support costs
• Reduce project risk of internally developed SIP applications

Features + Benefits

• UA, B2BUA
• Call Stateful Proxy, Transaction Stateful Proxy, Stateless Proxy, Redirect, and Registrar
• Transport
  - Meets IMS protocol specifications for SIP transport and transport security
  - UDP, TCP, SCTP
  - TLS, IPSec
  - IPv4, IPv6
• Fully multi-threaded Trillium protocol software providing optimal performance scalability on multi-core and multi-threaded processing architectures.
• Fault tolerant and highly available software based on Trilliums patented FT/HA framework.
• Lazy Parsing
• Extensive Testing:
  - Trillium XML Test Architecture (XTA) (1571 test cases)
  - Protos Denial of Service test suite (4800 test cases)
  - 3rd Party Test Box test suite (901 test cases)
  - SIPit Interoperability Testing
  - RFC 4475 SIP Torture Testing
  - ETSI TS 102 027 Conformance Testing for SIP
• Extensions:
  - Support for over 40 SIP specifications and extensions
  - Including 3GPP (IMS protocol), Instant Messaging, Presence, Conferencing, and Voicemail extensions
  - See Conformance below for a full-list of extensions
• Stack implementation features:
  - Seamlessly handles unknown methods and unknown headers.
  - Optionally can pass SDP as a string to the user for efficient implementations at the call agents.
  - Supports inherent call distribution to cater to scalable, carrier-grade applications.
  - Maintains various caches, required for efficient protocol implementation.
  - Supports the automatic generation of a number of SIP headers in the stack implementation.
  - Supports ordered and unordered message delivery over SCTP
• Database features:
  - Implements the cache database maintained using a multi-way radix tree implementation, optimized for memory usage.
• NS & UA features:
  - Sample NS, UA, and Stack Manager (SM) applications.
  - Supports APIs to contact the external/third-party registrar server and location server for address lookups.
  - Supports recursing on redirect responses received from a downstream server with minimal interaction with the call control application (UA only).
  - Acts as a standalone registrar, proxy, or redirect server with minimal interaction with the service user applications.
  - Acts as a standalone registrar server (without an application).
  - Supports automatic switchover from a stateful proxy to a stateless proxy upon depletion of available resources.
  - Supports call forking and can recurse on redirect responses received from a downstream server with minimal or no interaction with the network server application (NS only).
  - Addresses directed pickup cases (NS only).
  - Maintains various caches such as location results returned from an external location service or registrations learned by snooping on multicast addresses. These caches help expedite lookups and reduce customer protocol integration effort of customers - enabling them to concentrate on developing the call control application.
  - Implements the registry database using a multi-way radix tree implementation, optimized for memory usage.
• Conforms to Trillium Advanced Portability Architecture (TAPA)
• Additional Trillium software licensing benefits

Trillium SIP Block Diagram

Trillium SIP FT/HA Diagram

Product Interworking

Trillium SIP runs on top and is pre-integrated with Trillium TCP/UDP Convergence Layer (TUCL) and Trillium Stream Control Transmission Protocol (SCTP).

Trillium SIP interworks with Trillium SIP – Protocol Specific Function (PSF) to provide deployment proven fault-tolerance / high availability.

Trillium SIP is complemented by Continuous Computing's full line of Trillium VoIP and IMS protocol software, as well our AdvancedTCA and CompactPCI hardware platforms.

Professional Services

Continuous Computing also offers a wide range of onsite and offsite Trillium Professional Services including, but not limited to, software integration, application development, and protocol customization.

Conformance

Trillium SIP software supports the following standards:

• Core Specifications
  - RFC 3261 SIP
  - RFC 3263 Locating SIP Servers
  - RFC 3264 Offer/Answer Model with Session Description
  - RFC 3265 SIP Event Notification
  - RFC 3325 Asserted Identity Between Trusted Networks
  - RFC 3327 Registering Non-Adjacent Contacts
  - RFC 3581 Symmetric Response Routing
  - RFC 3840 Indicating UA Capabilities
  - RFC 3841 Caller Preferences for the SIP
  - RFC 4320 Addressing Non-INVITE Specification Issues
  - draft-ietf-sipping-config-framework-07 – A Framework for SIP UA Profile Delivery
  - RFC 3388 Grouping of Media Lines
  - RFC 4566 Session Description Protocol (SDP) – latest SDP specification
• PSTN Interworking
  - RFC 2848 PINT
  - RFC 3372 SIP-T
  - RFC 3398 ISUP to SIP Mapping
  - RFC 3204 MIME media type for ISUP and QSIG
• General Purpose Infrastructure Enhancements
  - RFC 3262 Reliability of Provision Responses in SIP (PRACK)
  - RFC 3323 Privacy Mechanism
  - RFC 3311 UPDATE Method
  - RFC 2976 INFO Method
  - RFC 3326 Reason header field
  - RFC 3420 Internet media type/sipfrag
  - RFC 3608 Extension header field for Service Route Discovery during registration
  - RFC 4028 Session Timers
  - RFC 2960 Stream Control Transmission Protocol (SCTP)
  - RFC 4168 SCTP as a transport for SIP
  - RFC 2915 The NAPTR DNS Resource Record
  - ITU-T(CCITT) Recommendation T.38 - Procedures for real-time Group 3 facsimile communication over IP networks.
  - RFC 3556 SDP Bandwidth Modifiers for RTCP
  - RFC 4244 Request for History Information
• Tel URL and ENUM
  - RFC 2806 URLs for Telephone Calls
  - draft-yu-tel-url-09 - Extension to “Tel” URL to support Number Portability and Freephone Service
  - RFC 3824 Using ENUM for SIP Applications
  - RFC 3761 - The E.164 to Uniform Resource Identifiers (URI) Dynamic Delegation Discovery System (DDDS) Application (ENUM
• Call Control Primitives
  - RFC 3515 REFER Method
  - RFC 3911 JOIN Header
  - RFC 3891 Replaces Header
  - RFC 3892 Referred-By Mechanism
  - draft-ietf-sipping-cc-transfer-05.txt - SIP Call Control – Transfer
  - draft-diversion-header-08 - Extension for diversion header
• SIP Service URI
  - draft-ietf-simple-event-list-07 - Event Notification Extension for Resource Lists
• Conferencing
  - RFC 3911 JOIN Header
  - draft-milanidovic-sip-multiparty-ext-02 - SIP Extension for Multiparty Conferencing
  - draft-ietf-sipping-conference-package-12 - Event Package for Conference State
• IM and Presence
  - RFC 3428 Extension for Instant Messaging
  - RFC 3856 SIP Extensions for Presence
  - RFC 3857 A Watcher Information Event Template-Package (WINFO)
  - draft-ietf-simple-im-sdp-01 - SIP Instant Message Sessions
• Event Frameworks
  - RFC 3903 Extension for Event State Publication
  - RFC 3680 Event Package for Registrations
  - draft-ietf-simple-event-list-07 - Event Notification Extension for Resource Lists
  - draft-ietf-sipping-conference-package-12 - Event Package for Conference State
  - RFC 3842 Message Summary and Message Waiting Event Package (Voicemail)
  - RFC 3856 SIP Extensions for Presence
  - RFC 3857 A Watcher Information Event Template-Package (WINFO)
• 3GPP and Other Extensions
  - RFC 4083 3GPP Release 5 Requirements
  - RFC 3455 P-Header Extensions for 3GPP
  - RFC 4032 Update to SIP Preconditions Framework
• SIP Compression
  - RFC 3486 Compressing SIP
  - draft-ietf-rohc-sigcomp-sip-01 – Applying Sigcomp to SIP
• Quality of Service
  - RFC 3312 Integration of Resource Management and SIP
  - RFC 3313 Private Extensions for Media Authorization
  - RFC 3524 Mapping of Media Streams to Resource Reservations
  - draft-ietf-mmusic-sdp-qos-00 - Establishing QOS and Security Preconditions for SDP sessions
• Security
  - RFC 3329 Security Mechanism Agreement
  - RFC 3323 A Privacy mechanism for the Session Initiation Protocol
  - RFC 3310 Hypertext Transfer Protocol (HTTP) Digest Authentication Using Authentication and Key Agreement (AKA)
• Emergency Services
  - RFC 4411 Extending the Reason Header for Preemption
  - RFC 4412 Communications Resource Priority
• Testing
  - RFC 4475 SIP Torture Testing
  - ETSI TS 102 027 - Conformance Test Specification for SIP
  - Protos Denial of Service Test Suite
  - 3rd Party Test Box Testing
  - SIPit18 & SIPit20 Interoperability Testing

Due to the dynamic nature of SIP technology, the above conformance standards list is constantly evolving. Please refer to the Session Initiation Protocol (SIP) Functional Specification for detailed functionality and specification compliance information.