SigIMS Data Flow Diagram, Hardware Accelerator Mode

SigIMS Data Flow Diagram, Hardware RTP Streaming Mode

SigIMS Data Flow Diagram, Software Only Mode

Asterisk Software Architecture Diagram

SigIMS IMS Platform

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• Overview
• Applications
• Interoperability Tests
• Advantages
• Mobile VoIP Page
• How to Purchase

Overview

The SigIMS platform is a configurable, multi-server based system that supports the following functionality:

• Session Border Controller (SBC) and Media Resource Functions
• Conferencing
• PSTN Interfaces
• Hardware Based Transcoding
• Encryption

For high channel capacity, SigIMS applications can be enabled with Texas Instruments VoIP device hardware, such as the SigC5561-PCIe card.

SigIMS includes one or more of the following software components:

• Kamailio (formerly called OpenSER)
• rtpproxy
• Signalogic software, including DirectDSP and modifications to Kamailio and rtpproxy
• Asterisk¹
• pjsip ²

SigIMS may also include one or more of the following hardware components:

• SigC5561 PCIe hardware
• SigC641x PCIe hardware

Hardware components are optional, depending on per-server channel capacity requirements. In software-only configurations, SigIMS runs with 100% functionality, but reduced channel capacity.

Applications

• VoIP
• Mobile VoIP

Session Border Controller (SBC) and Media Resource Functions

SigIMS platform SBC and media resource features include:

• software-based transcoding, including including G729, GSM-AMR, EVRC, iLBC, G726, G711 App1 , G723, CVSD, MELPe
• software-based encryption and header compression
• hardware-based transcoding (SigC5561 and SigC641x cards), including G729, GSM-AMR, EVRC, iLBC, G711 App1
• hardware-based encryption and header compression
• hardware-based echo cancellation (up to 128 msec)
• "per route transcoding", with routes specified config files
• "on-demand transcoding", where endpoints not capable of certain transcoding types are made to appear so; i.e. transparent transcoding
• Tones and announcements
• RTP time-out and tear-down
• Codec enforcement

Conferencing

Conferencing is supported both in software-only and hardware acceleration modes. The SigC641x-PCIe card supports voice conferencing, ranging from 680 3-node conferences up to one 2048 node conference.

PSTN Interfaces

PSTN interfaces are supported one of two ways:

• via Asterisk using various T1/E1 cards made by Digium, Sangoma, etc.
• via the wanPTMC-256T3 card , which provides a fully channelized T3 line interface

Hardware Based Transcoding

The SigC5561-PCIe card

Encryption

Interoperability Tests

SigIMS has been tested with the following software and mobile client components:

• Voipswitch
• ALOE Systems softswitches
• various smart phones running pjsip, including Nokia C3, C1-01, C3, C5, C5-03, C6-01, E7-00, E63, E65, E71, E72, N8, N93, N95, X3, X6 16GB, 1661, 2330, 2680, 2730, 3710, 5320, 5530, 6350, 7020
• various PC softphones, including X-Lite, eyeBeam, and SJPhone

Carriers tested include:

• AT&T
• Tata
• T-Mobile

Advantages

Some of the SigIMS advantages:

• Linux server oriented. Proprietary hardware avoided when possible
• by supporting both "software-only" mode and hardware acceleration mode, no API changes are required when adding (or subtracting) acceleration hardware. The same APIs are used in both cases -- for example, the DSAssignPlatform() API which has options for x86, ARM, and Signalogic accelerator cards
• Free software only versions available
• Several levels of tech support are available, including monthly and annual, with well-defined Minor, Major, and Critical response actions

How to Purchase

NOTES

¹ Asterisk is an open source B2BUA.
² pjsip is an open source mobile client with SIP and RTP interfaces, supporting Nokia phones, HTC phones (Android), and iPhone.