Product Bulletin Index	Description
BladeWareVXML Interpreter	BladeWareVXML Interpreter is the first product of a comprehensive open-source VoiceXML browser product offering. BladeWareVXML is an open-source VoiceXML interpreter that follows the W3C VoiceXML 2.1 dialog markup language specification, enabling developers to take advantage of well-known Web technologies and tools when building voice applications. An interpreter, like BladeWareVXML, is the keystone component of any standards-compliant voice browser.
BladeWareVXML Interpreter Commercial Edition (CE)	BladeWareVXML Interpreter Commercial Edition (CE) is the commercial version of Commetrex’ BladeWareVXML Interpreter open-source offering. It is an enhanced version of OpenVXi, the most widely-used VoiceXML interpreter on the globe. The enhancements made will be of great interest to many of the OpenVXi users; especially those looking for significantly improved performance and strict adherence to the VoiceXML standard. Moreover, VoiceXML is the critical enabling technology in the industry's move to separate service logic and content from the telephony-specific network elements, moving it instead to Web servers. It is the first product of a comprehensive open-source VoiceXML browser product suite from Commetrex Corporation. BladeWareVXML follows the W3C VoiceXML 2.1 dialog markup language specification, enabling developers to take advantage of well-known Web technologies and tools when building voice applications. An interpreter, such as BladeWareVXML, is an important component of any standards-compliant voice browser.
BladeWare Fax2Email	BladeWare™ Fax-to-Email Media Server (Fax2Email) is a BladeWare sample application that directly receives incoming SIP-based calls. It extracts the called number for the inbound call and uses it to query a database to retrieve fax and e-mail options. If the database lookup succeeds, the fax operation is completed and the result e-mailed to the specified e-mail address using a built-in SMTP server. Fax2Email also maintains call detail, SNMP, and Web-based management interfaces.
BladeWare FMS	BladeWare™ Fax Media Server (FMS) is a stand-alone IP-based media server that provides the real-time fax send-receive functions required to support applications such as unified messaging, never-busy fax, and hosted enterprise fax services. BladeWare FMS terminates both T.38 fax relay and G.711 pass-through fax, the latter being required to interoperate with gateways that do not support T.38.
BladeWare	BladeWare is the SIP-based HMP media-server edition of Commetrex’ Open Telecommunications Framework® (OTF) architecture. OTF is a digital-media system framework that supports any media, any network, any signaling, and any media-processing-related application. And it’s completely open. This means there are SDKs to support development at any level, including, of course, the application level, but also system services, signaling, and media technologies.
Caller ID	The 60085 CID Transmitter allows the developer of analog access equipment to offer caller ID compatible with PTT specifications in much of the world. In some situations, the message delivery sequence must be coordinated with events outside the scope of the transmitter, such as loop reversals or ringing. The task that controls the CID Transmitter is aware of the timing of these events and provides a signal to the CID Transmitter to indicate when transmission must begin and end. For the off-hook transmission, the TAS (TE Alerting Signal) tone generation and timing are implemented by the CID transmitter. When needed, the Channel Seizure Signal (CSS) and Mark Signal will be generated by the CID transmitter.
Call Classifier	The Commetrex Call Classifier (CC) determines whether a call is voice, fax, or data by examining the call’s media stream. This function is useful in media gateways that accept PSTN calls without knowing the type of stream processing, such as vocoder, fax or data modem relay, to place in the call path.
CPA - Call-Progress Analysis Licensed Media Technology	60080 CPA uses a low- and high-pass frequency discriminator together with energy measurements over time to qualify in-band-signaling tones. Three signal characteristics are used for tone detection: the tone frequency, timing (Tone_On, Tone_Off) and period and consistency (the number of times a tone cycle repeats before being qualified). The design of Commetrex’ CPA is based on these characteristics of call-progress (CP) tones: Nearly all CP tones are comprised of one or two frequencies between 200 and 500 Hz. Although the frequency and timing of call-progress tones varies between manufacturers and PTT administrations, there is a nearly universal signal-timing hierarchy that can be relied upon for discrimination between the various signals.
Fax Modem Bundle (Release 4.3)	Commetrex’ 60112 Fax Modem Bundle includes Commetrex’ V.21, V.27ter, V.29, and V.17 in a C-coded reference design and versions optimized for the Texas Instruments TMS320C6200 and TMS320C5400 DSPs in a TI XDAIS Algorithm Standard-conformant and Commetrex format. The optimized versions provide up to 16 simultaneous V.17 receivers for each 100-MCPS on a TI ‘C6400 or 7 simultaneous modems on the ‘C5400.
FaxTap PCM to TIFF-F Converter	FaxTap is an off-line (non-real-time) software process that converts the recorded PCM file to a TIFF-F file. In this case, non-real time doesn’t mean slow. A fax image that might take one minute to receive will be converted in 1-4 seconds. The function of the system is to convert a linear pulse-code modulation (PCM) recording of a facsimile session into the original transmitted image.
G.168 Line Echo Canceller	Commetrex’ G.168 Line Echo Canceller (LEC) allows the developer of telephony-endpoint products and gateways to meet the specification and speech-performance requirements of demanding enterprise and carrier applications. The LEC is available in C-reference source code and in highly optimized versions for the Texas Instruments TMS320C6000 line of DSPs. The LEC only requires 5.0 MCPS to execute on the ‘C6400, yielding a density of 120 channels per DSP with a 128-millisecond echo tail cancellation. Higher densities can be achieved with shorter tail lengths.
G.711 Licensed Media-Processing Technology	Commetrex' Media Processing Technologies include G.711, an ITU standards-based vocoder that addresses voice compression schemes. G.711 implements the standard mA-law compression/decompression schemes for encoding voice-frequency signals. Voice communication over the public switched and private (PBX) telephone networks is achieved at an 8000-Hz sampling rate. These samples are encoded as binary words and transmitted serially. Waveform coding schemes are designed to reproduce the waveform output of the source at the destination with as little distortion as possible. Pulse-code modulation (PCM) is the simplest and oldest waveform-coding scheme. The two common non-uniform PCM schemes that are used in voice applications are the m-law and A-law. In these techniques, the samples are passed through a nonlinear operation that compresses large amplitudes (reduces the dynamic range of the signal) and then performs a uniform quantization on the output. G.711 is among the ITU standards that address voice compression schemes.
G.723.1 - Dual-Rate Speech Coder for the TMS320C6000	Commetrex’ 60094-2 speech coder is a resource-efficient implementation of the ITU G.723.1 on the Texas Instruments TMS32C6000 family of DSPs. G.723.1 is widely deployed in H.324 conferencing systems and PSTN-IP gateways where high-quality speech and low-bandwidth utilization are requirements.
G.726 Licensed Media-Processing Technology	ITU G.726 is the most used speech-compression algorithm in telecommunications due to its high perceived speech quality and low resource requirements. It compresses 64-K speech to 40-K, 32-K, 24-K, or 16-K BPS using adaptive differential pulse-code modulation (ADPCM). Mean Opinion Scores (MOS) for G.726 at the 32K rate are considered to be toll quality; the 24K rate is near toll quality. G.726 is used in applications where high-quality, low DSP MIPS, and low algorithmic delay are desired. Since G.726 is a waveform coder, rather than a speech coder, it is more suitable in coding non-speech signals, such as DTMF, than some speech-specific coders. Commetrex offers G.726 in a C-coded reference design, as well as versions optimized for the Texas Instruments TMS320C6000 and TMS320C5400. Commetrex’ implementations are extremely resource-frugal. The ‘C6000 version requires only 1.55 mega-cycles per second in the dual-channel version where two samples are processed simultaneously. On the 200-MHz TI ‘C6201, this yields 120 channels per DSP. When the execution is optimized for single-channel operation the ‘C6000 version requires 2.77-MCPS, or over 60 channels per DSP.
G.729a/b - 8 KBPS CS-ACELP speech codec for the TMS320C6000	Commetrex’ 60095-2 speech codec is a resource-efficient implementation of the ITU G.729a/b on the Texas Instruments TMS32C6000 family of DSPs. G.729a/b is widely deployed in conferencing systems and PSTN-IP gateways where high-quality speech and low-bandwidth utilization are requirements. G.729 specifies a coding scheme that compresses 64-KBPS PCM speech to 8-KBPS using a conjugate structure algebraic-code-excited linear-prediction (CS-ACELP) algorithm. The coder is designed to process frames of 80 samples, or 10-milliseconds of speech data. A 5-millisecond “look ahead” is required, so the total delay is 15-milliseconds. G.729A is a reduced-complexity version of G.729. Since complexity reduction is due to internal computational simplifications, G.729 and G.729 Annex A are bit-by-bit interoperable. 60095-2 also includes the voice-activity detection and silence removal (VAD) and comfort-noise generation (CNG) specified in G.729 Annex B, hence G.729a/b.
HMPturbo™	Powerful servers and IP telephony have given rise to the host-media-processing (HMP) media server. The term “HMP” has been loosely applied by marketers since not all HMP media servers actually process a call’s media stream. The IETF RFC2833, which specifies how in-band telephony tones can be transported in RTP packets, allows the signal processing needed to detect these tones to be outsourced to the media gateway. And messaging can be implemented by simply storing the RTP packet payload. HMPturbo Transcoding Subsystem (TSS) is available in two versions: one requires no additional hardware. It’s for the processing-challenged media server. Here, HMPturbo is needed to meet additional functional requirements. Packet buffers are sent to the HMPturbo all-software subsystem through an easy-to-use API. The subsystem can handle, for example, in-band call-progress analysis, DTMF, real-time IP fax (G.711 or T.38) termination (and relay), and vocoder transcoding. The second version is functionally identical and exposes the same API, but uses Commetrex’ MSP-320 PCI care to really “turn on the turbo.” The same media-processing functions are available, but they are executed by the PCI board’s DSPs.
Image Conversion Library (ICL)	The Commetrex Image Conversion Library (ICL) is a stand-alone software package that supports the real-time conversions required by Commetrex’ PortableT.30 and a set of off-line conversions that support the creation of complex fax-image files. ICL’s support for dynamic re-encoding and page-width resizing means that T.30 connect negotiations will not fail due to a mismatch in facilities between the sending and receiving terminals.
MSP-320 Media Gateway	The Commetrex MSP-320 Media Gateway is an open-architecture hardware platform for the implementation of high-capacity integrated-media PC-based communications systems. The MSP-320, capable of processing 512 PCM streams, includes the resources to support all the media-processing needs of the modern enterprise communications server: high-speed data and fax, Internet telephony, voice processing for messaging, video processing, text-to-speech, and speech recognition.
MSP-320 Board-LevelDeveloper’s Kit	The MSP-320 is an H.100/PCI DSP board designed to give the developer an unprecedented array of options. For computer telephony (CT) applications, it can be used in a comprehensive hardware-software environment which includes the board, board-level software, drivers, and CT middleware, such as Commetrex’s Open Telecommunications Framework (OTF). The MSP-320 is ideal as a platform for multiline integrated-media stream processing using the MSP Consortium M.100-compliant OpenMedia streams environment. But should the developer require a proprietary board-level environment, the MSP-320 can provide the hardware foundation for use with this Board-Level Developer’s Kit to support the development and market deployment of proprietary board-level environments and media-processing software.
MSP-H8 Eight-Line PCM Interface	The MSP-H8 Media Gateway is a full-length PCI board with 8 high-performance analog trunk interfaces. A TI TMS320C5402 DSP is included for use as a control or media processor. When used as a control processor media processing is performed on the host. The MSP-H8 offers the ultimate in per-port affordability by taking advantage of the high MIPS ratings of PCs, eliminating the need for DSPs on the telephony interface board. The MSP-H8 includes up to 8 pluggable analog trunk interfaces with high-resolution 16-bit linear codecs and the low-distortion characteristics needed for high-speed data modems. The PCM data are buffered and transferred to the host PC via an efficient PCI plug-and-play interface.
MSP-H8 Embedded SDK	The MSP-H8 is an 8-line PCM interface PCI board that supports both embedded and host-level signal-processing applications. The Embedded SDK gives the developer all the tools needed for the productive development of the call-stream processing required by applications such as terminating voice, fax, and data, as well as PSTN-IP gateway and IAD applications. The MSP-H8 is ideal as a platform for low-density multi-line integrated-media stream processing using the Commetrex’ OpenMedia streams environment. But, should the developer require a proprietary board-level environment, the MSP-H8 offers the hardware foundation for use with this Embedded Developer’s Kit to support the development and market deployment of proprietary board-level environments and media-processing software.
OpenEndpoint	Commetrex’ OpenEndpoint provides the telephony software required to implement an IP endpoint voice-fax gateway or integrated access device. A comprehensive product, OpenEndpoint includes POTS interface software, signal generation-detection, packet voice, and T.38 and/or I.366.2 fax relay executing in Commetrex’ OpenMedia standards-based streams framework. All the OEM need add for a complete IP endpoint product is the signaling and application software. SIP and H.323 are optionally available. Although the software is available in portable-C format, a highly optimized version is available for the Texas Instruments TMS320C6000 family of high-performance DSPs. A single low-cost 200-MHz ‘C6000 provides, for example, 12 non-blocking channels running G.723.1. The TMS320C6400 can support 120 channels running G.726 and a G.168-compliant echo canceller with a 16-milli-second tail. Versions supporting the ‘C5000 are also available.
OpenMedia™ SDK	OpenMedia™ is an open multi-stream, media-processing software environment that supports IP media-stream processing for voice-over-IP (VoIP) and fax-over-IP (FoIP), as well as traditional computer telephony circuit-switched applications. OpenMedia reduces the cost of developing and porting media-processing products and supports straightforward media software deployment on Commetrex’ MSP boards, host PCs, and other hardware platforms. OpenMedia can be seamlessly integrated with the Open Telecommunications Framework (OTF) Kernel, Commetrex’ telephony middleware, as well as other vendors’ software environments.
OTF for MSP	Commetrex has decomposed the generic digital-media telephony system into value-adding system elements that complement virtually any development strategy: OTF Kernel is the industry’s only vendor- and resource-independent telephony middleware. OpenMedia is the only licensed standards-based streams framework. MSP (Media Stream Processor) is Commetrex’ line of call-stream-processing resources. OTF for MSP is the combination of these individual products into an integrated platform for the application-level system developer. With OTF for MSP, the developer benefits from the productivity of developing on a full-function platform, but without giving up the option of adding proprietary functionality at the middleware, media-resource, or algorithm level.
Open Telecommunications Framework Kernel (OTF Kernel)	Open Telecommunications Framework Kernel (OTF Kernel) is a modular, distributed, client-server, open implementation of the ECTF S.100 R2 recommendation (see http://www.ectf.org). It is a vendor- and hardware-independent telephony-middleware system kernel that provides the core S.100 system services in the Win32, Linux, and Sun Solaris environments. OTF Kernel is used by developers of access equipment, gateways, service platforms, and enterprise equipment when complete control of the product platform is required yet time-to-market and investment hurdles preclude internal development of the system framework, which often requires the greatest investment of any major component in a telephony system.
PortableT30	There are four major elements of a typical fax application: fax modems, T.30 protocol engine, image-conversion and management, and the application. If every fax transaction is successful the factor that leads most directly to customer satisfaction is the application. But there is nothing an effectively designed and implemented application can do to overcome poor performance of the underlying fax technology, and the system element most responsible for high connect rates, the biggest term in the performance equation, is the T.30 fax protocol engine. Commetrex’ PortableT30TM, with Always Connect Technologyä, provides the highest connect rates in the industry, reflecting its 7 years of field deployment.
PowerCall for OTF Kernel	OTF Kernel, Commetrex’ telephony middleware, is licensed without media-processing resources. PowerCall for OTF Kernel adds signal generation-detection capability and in-band call-progress analysis to an OTF Kernel system. The PowerCall OTF Kernel SDK provides the necessary development environment; runtime licenses add a specific number of runtime resources to an OTF system.
PowerFax – Fax Send-Receive Resource for OTF	PowerFax for OTF is the most flexible multi-line fax-server resource ever developed and marketed. OTF Kernel, Commetrex’ S.100 middleware, is licensed without media-processing resources. PowerFaxÒ for OTF adds Group 3 fax send and receive capability to an OTF system. The PowerFax for OTF SDK provides the necessary development environment; runtime licenses add a specific number of runtime resources to an OTF system. These runtime resources can be for any MSP-series hardware resource, such as the MSP-H8 analog interface and the MSP-320 quad-span DSP resource, or, the optionally available PowerFax TerminatingT38, which supports fax send-receive in real time over IP networks. Moreover, Multi-Modal Terminating Fax (MMTF) adds G.711 packet-based modem capability to deal with gateways that do not offer T.38 support.
PowerRelay	Commetrex’s real-time fax relay technology gives the next-generation network system developer the technology needed to support standards-based fax in IP, ATM, and Frame Relay networks. As a developer of a packet-telephony system you can offer a T.38-compliant system for fax over IP, an I.366.2 system for fax over ATM, an FRF.11.1-compliant system for fax over Frame Relay, or a proprietary implementation, should it be required.
PowerVox – Voice Play-Record for OTF Kernel	OTF Kernel is licensed without media-processing resources. PowerVox for OTF Kernel adds voice play and record capability to an Open Telecommunications Framework® system. The PowerVox OTF Kernel SDK provides the necessary development environment; runtime licenses add a specific number of per-call resources. These runtime resources can be for any MSP-series resource hardware or the terminating version of any voice coder that supports voice-over-packet networks. PowerVox for OTF Kernel provides the OTF developer an ECTF S.100-conforming Player-Recorder API, allowing the client application to access PowerVox’s functionality through an open-standard API. The resulting application is portable across the available S.100 server implementations.
MSP-320 Quad E1/T1 Daughterboard™	The Commetrex MSP-320 Quad E1/T1 Daughterboard™ adds four E-1 or T-1 interfaces to the Commetrex MSP-320. The board includes four line interfaces and framers controlled by an on-board Intel 386EX co-processor. The Daughterboard-main board interface includes 256 PCM timeslots. The daughter board eliminates the need for a separate network-interface board and takes the MSP-320 to quad-span densities.
Signal Generator/Detector (SGD)	Commetrex’ Signal Generator/Detector (SGD) furnishes all necessary functionality at the physical-layer level to support in-band telephony-signaling functions. Specifically, the SGD implements signal-processing functionality to generate and detect DTMF and Multi-frequency (MF) (R1/R2) digits as well as any generic in-band tone comprised of up to two frequencies. The SGD is compliant with the MSP Consortium M.100 recommendation, the Texas Instruments XDAIS algorithm packaging standard, the ECTF S.100 requirements for signal generation and detection, and the ITU Q.24 recommendation.
TerminatingT38	Commetrex’ TerminatingT38™, a special combination of its ITU T.38 and T.30 ITU fax protocol engines, gives the developer of a network-service platform or an enterprise fax server the technology necessary to terminate T.38 (IP network) real-time fax transmissions the same as it would real-time faxes from the PSTN using analog modems. When combined with Commetrex’ stand-alone T.38, the developer can implement both terminating and standard fax-gateway functions on the same platform, combining the functions of the service platform and the gateway. When combined with Commetrex’ fax modems the server’s application can send and receive faxes without regard to whether they are transported via TDM or packet networks, or gateways with and without T.38 support.
Timescale Modification MST	There is a need in voice-messaging systems for a feature that allows the user to control the speed of message playback. This capability is generically called “time-scale modification” or TSM within the speech-processing industry. There are many methods of implementing this capability, but the best of them allow the subscriber to stop and start playback as well as to slow down or speed up the message playback by a factor of from 0.5 (half speed) to 1.5 (1.5 times normal speed) while maintaining the fidelity of the original recording played at the original real-time speed (normal speed). The user usually controls the speed by pressing DTMF digits on a telephone handset, ASR (Automatic Speech Recognition) voice commands, or pointing devices on computer-based terminals.
Voice Transcoding Subsystem (VTS)	The Voice Transcoding Subsystem (VTS) is a software product, typically used in circuit-switched trunking applications, which implements T1 and E1 transcoding as specified in the ANSI T1.302a-1992 and ITU-T G.761 specifications. VTS compresses two T1 or E1 lines into a single line or decompresses a single span into two. Compression is accomplished using G.726. Although it can be adapted to other processors, VTS is specifically optimized for the Texas Instruments TMS320C6000 DSP. In full-duplex mode it operates on a pair of TI TMS320C6211-150 processors, or on a single more-powerful member of the ‘C6000 product family.