NMS Communication - Fusion 4.2

IP Telephony Development Platform - Featuring MGCP Support

NMS Communications' Fusion™ is the industry's most scalable, highest performance development platform for standards-based Internet Protocol (IP) telephony solutions. Fusion 4 supports both PCI and CompactPCI platforms, and provides a common software development environment which can be used to create IP telephony gateways, IP-enabled enhanced services platforms, and wireless IP telephony gateways. The Fusion 4 modular architecture allows support for standard protocols such as the International Telecommunication Union's (ITU) H.323 specification, the Media Gateway Control Protocol (MGCP), the MGCP-based Megaco (ITU's H.248), and the Internet Engineering Task Force's (IETF) Session Initiation Protocol (SIP). Fusion 4 has a scalable architecture which enables developers to create application solutions with configurations from 120 ports (bi-directional voice conversations) to over 1000 ports with no increase in latency or decrease in performance. Fusion 4 runs on top of NMS' acclaimed Natural Access™ development and runtime environment. The rich set of high-level APIs allows easy application development and reduced time to market.

Fusion 4 uses an intelligent hardware and software architecture that integrates PSTN interfaces, telecommunications protocols, comprehensive IVR functionality, full-duplex echo cancellation, speech encoding, fax processing, LAN interfaces, and data protocols into a cohesive, flexible package. The Fusion 4 software development kit (SDK) includes reference code and sample applications that can be used with minimal configuration to quickly provide proof-of-concept for developers. The reference code is a time-saving programming tutorial, enabling developers to concentrate on adding the features and functions that differentiate their IP telephony solutions from other market offerings.


  • Multiple product configurations, providing high scalability that delivers the best price/performance platform available
  • Industry-leading power efficiency, allowing realistic creation of high-density configurations of over 1000 ports per chassis
  • High-performance, low-latency architecture that does not degrade as system scales
  • Single, powerful software development environment across product family, simplifying application creation
  • Media Gateway Control Protocol (MGCP) support offering:
    • Compliance with RFC 2705
    • Conformance to International Softswitch Consortium's MGCP Implementer's Guide
    • Support for basic MGCP packages — generic media, RTP, line, trunk, MF, DTMF, and announcement server
    • Ability for application to implement MGCP extension packages to extend service capabilities
    • Fully integrated Natural Access service, complete with trace capabilities
  • Support for the ITU H.323 specification, enabling interoperability with other H.323-compliant clients, gateways, and gatekeepers
  • Capability to support future control protocols — Megaco (H.248) and SIP
  • Broadest choice of standard vocoding algorithms, including G.723.1, G.726, G.729A/B (with VAD), and G.711
  • Continuing vocoder development to add ETSI GSM FR and EFR, G.728, AMR, and others
  • Real-time (T.38) and store-and-forward (T.37) fax on the same platform in a universal port model
  • Supports T.38 fax standby channel for easy voice to fax switchover on same channel
  • Support for multiple operating systems for maximum flexibility — Windows® , Intel® and SPARC® Solaris™ , and Linux® (MGCP supported only by Windows NT and SPARC Solaris)
  • Full IVR support for voice front-end capability, including two-stage dialing, standards-based in-band DTMF carriage, and out-of-band DTMF
  • Support for on-board conferencing of PSTN and IP-based audio streams
  • On-board implementation of standard Internet protocols, including the User Datagram Protocol (UDP) and Real Time Protocol/Real Time Control Protocol (RTP/RTCP), allowing maximum performance and scalability
  • Duplication and re-direction of RTP streams, including forking, to conform to CALEA requirements
  • Ethernet link and IP forwarding status available to applications
  • Clarent ThroughPacket™ combines voice packets from several calls into one complex packet with a single header overhead, thus reducing both the bandwidth required and the level of packet congestion in a network
  • Compact footprint supporting up to 360 ports of IP telephony capability in a single cPCI slot
  • Switching fabric based on industry-standard H.100/H.110 bus (CT Bus) as a switching fabric, easing integration with other H.100/H.110-compliant products
  • Ideal for toll bypass, voice and fax messaging, web-enabled call centers, IP-only enhanced services, virtual second line, and wireless gateway applications

Product Description:

Fusion 4 integrates hardware and software within a standard PCI or CompactPCI (cPCI) computing environment, greatly simplifying development and deployment of IP telephony gateways. Fusion's field-proven hardware components and industry-leading Application Programming Interfaces (APIs) minimize programming requirements and maximize flexibility.

Fusion 4 Hardware

Fusion 4 runs on the CG 6000 Series. and the CG 6500C. The CG 6000/CG 6500C Series provides PSTN interfaces, call control and protocol support, and universal port, real-time DSP functionality. Standard universal port Fusion functionality includes vocoding with echo cancellation and real-time fax (T.38). The CG 6000/CG 6500C Series also provides a dedicated packet-processing engine with Fast Ethernet interfaces.

Fusion 4 makes use of the CG 6000/CG 6500C Series' complete worldwide protocol support and approvals. The CG 6500C delivers up to 360 ports of standard universal port Fusion capability in a single slot. Multiple CG 6000/CG 6500Cs may be cascaded in a single chassis to build high-density solutions.

The CG 6000/CG 6500C Series provides dual Fast Ethernet interfaces and a dedicated IP and RTP packet-processing engine, which delivers from 12,000 to over 20,000 packet/sec performance for the CG 6000 and up to 60,000 packet/sec for the CG 6500C.

Recommended Configurations:

Configuration Fusion Ports
cPCI chassis CG 6500C-{0 or 2}L/{0, 8, or 16}4TE 360
cPCI chassis CG 6000C-2L/4TE 120
PCI chassis CG 6000-2L/{0,2, or 4}TE 120

Fusion 4 Software

The Fusion 4 software development kit consists of the following components:

  • Natural Access 4 (NACD 2001-1 or later) for comprehensive support of the CG 6000/CG 6500C family, providing:
    • Call control
    • Voice processing
    • Switching services
    • DSP control
    • Operations, Administration and Management (OAM) services
  • IP Network APIs
    • CG 6000/CG 6500C Series loading and control
    • Direct RTP/RTCP control
    • H.323 or MGCP API for IP call control (optional)
  • Fusion 4 Vocoder Media Kits
  • Fusion 4 Gateway Reference Sample Code
    • Full source code of reference nailed-up gateway application
    • Prototype application exercising Fusion APIs
    • Jump-start on proof-of-concept demos and prototyping

Technical Description:

When Fusion 4 applications receive incoming calls, they spawn caller threads and use Natural Access to perform the following tasks: application initialization, port initialization, call control, event processing and error handling, and parameter management. Additionally, Natural Access' Switch Service provides a way of making, breaking, and controlling the H.100/H.110 connections between boards.

Fusion's software architecture runs RTP/RTCP on the family of CG 6000 platforms and keeps the remaining IP control protocol functionality on the host, relieving the host from processing real-time audio packets.

Fusion 4 introduces Media Stream Packet Processing (MSPP) as an API for building media streams. An MSPP connection consists of two endpoints and a channel to connect them. There are different types of endpoints depending on the external connection: DS0 endpoints support pulse code modulation (PCM) voice; RTP endpoints support VoIP data; TPKT endpoints support Clarent Throughpacket operations; and T38UDP endpoints support T.38 fax data. The application builds channels that contain filters that operate on the data as it flows through the channel. A full-duplex MSPP voice channel, for example, is made up of 4 filters: jitter buffer, bridge, voice decoder, and voice encoder.

Universal Port DSP
Fusion 4 supports a universal port DSP model, in which each port is able to handle call control, voice processing, vocoding with echo cancellation, and fax processing simultaneously. A broad range of vocoders is supported for maximum flexibility of gateway deployment. Both Fusion's real-time (T.38) and Natural Access' store-and-forward (T.37) fax are available — a unique capability in the industry. In response to the rapid evolution of standards and DSP technology, Fusion also provides an open DSP platform for easy porting of other algorithms as they gain market share, are approved as standards, or are otherwise required by customers.

Media Gateway Control Protocol
Media Gateway Control Protocol (MGCP) enables a VoIP gateway to be decomposed into a call control component (call agent), a signaling component (signaling gateway), and a media control component (media gateway). MGCP is the protocol used between the call agent and the media gateway. The media gateway converts audio signals between the circuit-switched network (PSTN) and the IP-based packet network under the direction of the call agent. Other VoIP entities that can use MGCP are an application server and a media server. The media server is a specialized IP-only device that provides enhanced services (such as voice mail, announcements or unified messaging) for VoIP and needs no circuit-switched interface.

The MGCP Service provides a simple API of eleven functions and enables rapid development of media gateways or media servers, including configuration and extension package request and response functionality.

To implement a basic VoIP trunking gateway, the application need only configure the number and type of MGCP endpoints, utilizing only two API functions. The gateway is then controlled by a call agent (sometimes known as a softswitch or media gateway controller), without any further application responsibilities. This allows application developers to focus on implementing enhanced functionality through extensions to MGCP.

MGCP may be extended by defining new "packages." A package defines a set of related signals and events that define some capability. Thus, the capabilities of an MGCP endpoint are determined by which packages it supports. Signals, such as caller ID, DTMF digits, or announcements, are generated by the endpoint. Events, such as Off-hook, DTMF digits, or tones, are detected by the endpoint.

MGCP Service integrates a number of other Natural Access Services in order to act on the commands from the call agent without requiring application intervention. All calls to MSPP, ADI, SWI, and VCE are performed internally within the MGCP Service. A key feature of NMS' MGCP Service is that it allows the application the flexibility to define new packages. These new packages are implemented by the application utilizing other Natural Access APIs. Applications may implement enhanced capabilities that are beyond the basic capabilities provided — for example an extended play and record or conferencing functionality. In these cases, the application needs to implement the events and signals of the extension package. But communication with the call agent, timeslot switching, and MSPP function calls are still made by the MGCP Service.

H.323 Protocol
H.323 is a broad standard from the ITU that sets specifications for audio, video, and data communications over IP-based networks. Additionally, H.323 specifies a series of vocoders to guarantee interop-erability among gateways and clients from different vendors.

NMS offers a field-proven H.323 stack from our partner RADVISION. Fusion's H.323 support includes H.225 and H.245. The H.225 standard specifies the syntax and semantics for negotiation at the start and/or during communication. H.245 specifies media packetization and call setup. For applications that have unique protocol requirements, other stacks, such as SIP, may be easily substituted into Fusion's software architecture.

RTP and RTCP are the accepted standards for passing real-time data streams over an IP network. Many higher level protocols, including H.323 and SIP, depend on RTP as an underlying transport mechanism. Fusion's RTP interface provides applications with low-level control over connections that pass real-time data between a circuit-switched network and an IP-based packet network. All RTP connections are initiated, monitored, and eventually terminated via the RTP interface.

RTCP allows an application to receive RTCP-related information. An RTCP monitor task may be developed for collecting RTCP statistical information on the host. This will allow for QoS monitoring during a session and provide a mechanism for collecting session-specific billing information.

Programmable Jitter Buffer
A unique feature of Fusion is its programmable jitter buffer. As voice packets are transferred across an IP network, packets may be lost or arrive out of sequence. A jitter buffer collects incoming packets and enables Fusion to rearrange the packets into the correct order or to smooth over lost packets. The size of the jitter buffer is configurable on a per channel/session basis, offering a unique feature to control latency for real-time, interactive voice conversations.

Product Specifications:

Host System Requirements

  • Windows NT, Windows 2000, and Linux
    • 128 MB RAM minimum
    • Pentium II 266 MHz or higher
  • SPARC Solaris
    • 256 MB RAM minimum
    • Sun SPARC 300 MHz or higher

Hardware configuration

See indivdual data sheets for specifications

Operating System Support

  • Windows NT 4.0 Service Pack 5 or greater
  • Windows 2000
  • Intel Solaris 7
  • SPARC Solaris 7 (32-bit)
  • SPARC Solaris 8 (64-bit)
  • Red Hat® Linux V6.2

Universal Port DSP

  • Comprehensive voice support
  • Vocoders: G.711 (both µ-law and A-law), G.723.1, G.726, G.729A/B
  • Vocoder features
    • On-line/off-line control
    • Gain control in dB
    • Payload ID allows application to use a non-standard name for vocoder
    • Compound packets
  • Real-time T.38 fax
    • UDP only
    • Fax ECM support
    • Packet compounding
    • Fax billing statistics
    • Packet redundancy
  • Echo cancellation
    • G.168 or equivalent
    • Up to 64 millisecond tails

Voice Encoder Controls

VAD (On/Off)

Rate (kbps)

Tones (On/Off)


Protocols Supported

  • IP
    • IP
    • UDP
    • RTP/RTCP
  • SNMP
    • RTP MIB
  • DTMF Carriage
    • H.245 (out-of-band)
    • RFC 2833 (in-band)
  • PSTN
    • Analog protocols
    • ISDN
    • Wink protocols
    • MFC-R2
    • CAS support of versions specific to many countries
  • Gateway control (runs on host)
    • H.323, including H.225, H.245, and Q.931
    • MGCP (Windows NT and SPARC Solaris only)