CLARENT GATEWAY 3.0
Clarent Corporation 1900 Broadway
Redwood City, CA 94063
Clarent Corporation, recently announced their Clarent Gateway 3.0, a new
high-port-density gateway designed specifically for both traditional and next-generation
Internet telephony companies. Clarent Gateway 3.0 is designed to handle up to 120 ports in
a single chassis four times the capacity of Clarents previous 2.0 release
all in the same footprint.
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
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.
Clarents 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.
ACCESS PLUS F200IP
Nuera Communications, Inc. 10445 Pacific Center Court
San Diego, CA 92121
Web Site: www.nuera.com
Since its release late last year, Nueras Access Plus F200ip voice FRAD/IP gateway
server has won its share of accolades. Were proud to announce that its
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. Nueras 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 Nueras gateways can be scaled up to support tens of
hundreds of T1/E1 PBX connections, and thousands of voice channels as packet voice
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.
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
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). Nueras 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 Nueras products call
619-625-2400 or visit the companys Web site at www.nuera.com.
DM3 IPLINK T.38 Real-time Fax
Dialogic Corporation 1515 Route Ten
Parsippany, NJ 07054
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
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
Dialogics 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
For more information, call Dialogic at 973-993-3000 or visit the companys Web
site at www.dialogic.com.