Second Quarter 1998

Enhancements in Digital Signal Processor (DSP) technology are providing faster, more powerful DSP engines, and so, Clarent has moved some of the voice processing functions from the host CPU to the DSPs. The result is faster processing power. The new generation of DSP processors provides much improved quality and throughput with the ability to support Internet data connections in addition to voice and fax calls. This increases the number of ports each gateway can support, which in turn lowers equipment costs without additional increase in rack space. Clarent estimates that the enhancements afforded by newer, more powerful DSPs will decrease the price per port of the Gateway 3.0 by as much as 35 percent. The new gateway is in beta trials and is scheduled for general availability in mid June.

The Clarent Gateway 3.0 incorporates the latest generation of DSP technology from Audio Codes and Phylon, as well as quad (four-channel) T1/E1 telephone interface technology from Natural Microsystems. The Gateway will also support the industry gateway standard G723.1, and G.729 compression/decompression (codec) algorithms.

LATENCY REDUCTIONS
 In other recent news, Clarent announced that they have further enhanced their already clear voice quality to provide true Public Switched Telephone Network (PSTN) quality over the Internet by attacking one of the major headaches faced in our industry — latency.

With enhanced voice processing technology, Clarent has reduced its already low Voice over IP latency — the time between when a word is spoken on one end of a phone call and when it is heard on the other end — by half. This technology development will enable Clarent to offer their carrier and Internet Telephony Service Provider (ITSP) customers a competitive edge by offering Internet telephony voice quality that is indistinguishable from that of PSTN-based calls.

Clarent’s new development has reduced the "mouth-to- ear" latency time of VoIP calls from 280 milliseconds (ms) to 150 ms, or roughly from one-third of a second to one-sixth of a second. This measure includes domestic PSTN access on both ends of the call. The time lapse is indistinguishable to the human ear and is comparable to the latency experienced on a PSTN phone call.

For more information, call the company at 650-306- 7511 or visit their Web site at www.clarent.com. 

Since its release late last year, Nuera’s Access Plus F200ip voice FRAD/IP gateway server has won its share of accolades. We’re proud to announce that it’s happened again. The product supports phone-to-phone connections over frame relay, Internet, and private IP networks. The F200ip also features E-CELP (Enhanced Code Excited Linear Predictive, a speech compression method that achieves high compression ratios along with toll quality audio) along with Ethernet-based IP transmission and serial frame relay trunks that support full T1 throughput. Nuera’s offering also offers software configurable analog voice interfaces, digital T1/E1 PBX interfaces, and capacity up to 30 voice/fax channels per unit.

The F200ip routes calls based on either user selection or system manager configuration. Ethernet support means that Nuera’s gateways can be scaled up to support tens of hundreds of T1/E1 PBX connections, and thousands of voice channels as packet voice applications grow.

The Access Plus F200ip provides flexibility in network protocols for optimizing quality/cost tradeoffs by running both voice over IP and voice over frame relay concurrently. Another advantage is that the product offers simple LAN installation and provisioning when using IP.

Getting back to the issue of voice compression, Nuera tells us that the Access Plus F200ip provides the highest quality and widest range of voice compression technology in the industry. Here are some of the details: • ITU G.728 LD-CELP, G.729 CSA-CELP, and G.726 ADPCM standard algorithms from 8 kbps to 32 kbps.

• E-CELP advanced proprietary algorithms operating at 4.8, 7.47, and 9.6 kbps.
• Low-delay voice encoding and adaptive jitter buffer minimizes end-to-end speech delay.
• Integral echo cancellation — adapts from 0-49 ms to ensure consistent voice quality over long tail circuits.
• Sophisticated lost packet recovery methods help maintain consistent quality in harsh environments.
• Voice compression rates negotiated to provide configuration flexibility.
• Modem transparency over voice channels.

The F200ip incorporates unique technology designed to minimize WAN bandwidth such as programmable voice packetizing (to improve bandwidth efficiency); adaptive silence suppression (minimizes bandwidth usage during speech breaks); and asymmetric fax channels (minimize return path bandwidth usage). Nuera’s Access Plus F200ip gateway is base priced at $4,195. Voice channel prices range from $1,000 per port for digital modules to $1,400 per port for analog modules. For more information on Nuera’s products call 619-625-2400 or visit the company’s Web site at www.nuera.com.

Dialogic recently announced an implementation of a real-time Internet fax standard. Dialogic has added T.38 (formerly known in the ITU as T.Ifax 2), a new real-time fax coder, to its DM3 IPLink line of products. DM3 IPLink is a complete, H.323 standards-based server development platform for building IP telephony solutions.

T.38 includes contributions from the United States and Israel. The initial proposal from RadLinx and follow-up work from Dialogic, Brooktrout, and Lucent, resulted in a working document that is slated for discussion and possible approval at the June, 1998, ITU meeting. While previous standards efforts focused on X.25 networks, T.38 is the first true standard for sending faxes in real time over IP networks. T.38 has been selected as the fax coder within the H.323 IP Telephony Call Control Standard, thus ensuring interoperability with other H.323-compliant devices.

Dialogic recently demonstrated the technology. The demo included two fully operational H.323 Internet telephony gateways. A voice phone call placed from the handset of a fax machine was connected over the IP network using a G.723.1 voice coder. When the fax transmission started, the H.323 gateway immediately detected the fax tone, selected the T.38 fax coder, and delivered the fax in real time. At the end of the transmission, the call was returned to voice mode. It works.

Because T.38 fax is handled within the H.323 framework as simply another coder type, anyone currently using the DM3 IPLink solution needs only a software upgrade to enhance their solutions to include fax support — no application modifications are necessary. Customers building IP telephony gateways for corporate and telco markets will be able to provide a universal port supporting real-time voice and real-time fax on the same port, and even on the same call.

DM3 IPLink products are standards-based, real-time IP voice and fax solutions built on Dialogic’s DM3 Mediastream architecture. The ultra-high density DM/IP3030c enables a 480-port IP telephony chassis (eight boards with sixty ports), which can be PCI-, CompactPCI-, or VME-based.

As a result of the T.38 implementation, Dialogic can offer a truly powerful solution for corporate and carrier- class IP telephony needs. DM3 provides a powerful platform to develop and deploy both IP telephony and messaging-based CT solutions. DM3 is scalable from the low-density, four-port analog solution up to the QuadSpan Series, providing 96 to 120 ports.

For more information, call Dialogic at 973-993-3000 or visit the company’s Web site at www.dialogic.com.