IPM-1610 [ETC]

16 E1/T1 cPCI VoIP Media Processing Board ; 16 E1 / T1的cPCI VoIP媒体处理板\n
IPM-1610
型号: IPM-1610
厂家: ETC    ETC
描述:

16 E1/T1 cPCI VoIP Media Processing Board
16 E1 / T1的cPCI VoIP媒体处理板\n

PC
文件: 总2页 (文件大小:77K)
中文:  中文翻译
下载:  下载PDF数据表文档文件
TM  
AudioCoded Enabling Technology Products  
IPM-1610  
16 E1/T1 cPCI VoIP Media Processing Board  
The IPM-1610 is a complete VoIP media processing solution providing IP and  
PSTN interfaces as needed to build next generation applications for both  
today's and tomorrow's networks. Moreover, introducing a packet interface  
for voice streaming the board enables a smooth evolution from legacy PCI  
bus architecture to distributed, scalable, packet-based architecture.  
Deliver Feature-Rich Solution  
A broad selection of firmware based media processing capabilities is available  
with the IPM-1610 including: message record/playback, conferencing,  
on-board announcements storage, voice coding, echo cancellation, fax  
processing and call progress tone detection. Each channel resource on the  
IPM-1610 is universal and can perform media processing functions while  
utilizing full flexibility in endpoints.  
Comply with Industry Standards  
The IPM-1610 board complies with industry standard network control protocols  
including MGCP, MEGACO (H.248), SIP or AudioCodes’ proprietary TPNCP.  
These allow for the implementation of a distributed media server architecture  
that separates call-processing functions from media processing functions,  
resulting in better redundancy, scalability and higher system availability.  
7 IP-enabled, cost-effective technology  
7 Field-proven PSTN interface board  
7 High-density, high performance board  
7 Independent call-by-call basis LBR ports  
7 Carrier grade applications  
Protect Customer Investment  
th  
The IPM-1610 is a member of AudioCodes’ 4 generation of the widely  
deployed TrunkPack Media Gateway on a blade family. Like the other members  
of the TrunkPack family the IPM-1610 supports AudioCodes API, which  
enables software download, provisioning and control. Maintaining essential  
API backward compatibility of future releases in order to protect customers'  
investment, is an important feature of AudioCodes' software update/upgrade  
program.  
7 Concurrent toll quality voice and fax support  
7 Enables scalable distributed architecture  
7 Shorter development cycle  
Enable Fast & Easy Integration  
Enabling accelerated design cycles with high-density and reduced costs, the  
IPM-1610 is an ideal building block for scalable, reliable VoIP enabled media  
processing solutions. With the IPM-1610’s comprehensive feature set,  
customers can quickly design a wide range of solutions combining PSTN and  
VoIP networks.  
IPM-1610 Features  
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Up to 480 IVR streaming ports  
Up to 240 Universal media processing ports  
Voice Record/Playback  
Fax termination and generation (T.37)  
Interchangeable RTP or PSTN or H.110 endpoints  
Real-time, multi-party conferencing  
Comprehensive IVR control  
VoIP packet streaming (RTP/RTCP) per RFC 1889/1890  
MGCP, MEGACO, SIP and AudioCodes' proprietary TPNCP  
cPSB PICMG 2.16 compliant Ethernet on the backplane  
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Automatic Speech Recognition (ASR)  
1
Text To Speech (TTS)  
TM  
AudioCoded Enabling Technology Products  
IPM-1610  
Specifications  
Applications  
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Call Centers  
Software Specifications  
Conference Servers  
IVR Servers  
Unified Communications/Messaging  
Voice Portals  
CTI Applications  
Voice Recording  
Configuration  
“Streaming”- 480 Voice/Fax Messaging ports  
“Universal” - 120, 240 universal ports  
Host-based record/play, WAV format (G.711, MS-GSM)  
Up to 240 ports of record/play - PCI bus based  
Up to 480 ports of record/play - packet streaming based  
Playback speed control with pitch correction  
Time Slot summation - Record RX+TX of the call  
On-board announcement storage - 10 Mb  
Recorded prompts - 20 minutes of G.711, 200 minutes of G.723  
Supports up to 240 ports of Mixed IP, PSTN and TDM (H.110) participants  
Maximum half-duplex parties per conference bridge: 120  
Maximum simultaneous 3-way conferences per board: 80  
Maximum full-duplex parties per conference bridge: 64 endpoints  
Supports various conference control modes  
Termination/Generation  
Voice Messaging, Recording  
About AudioCodes  
AudioCodes Ltd. (NASDAQ: AUDC) designs,  
develops and markets Voice over Packet media  
gateway technologies and systems for  
converged networks. The company is a market  
leader in voice compression technology and  
the key originator of the ITU G.723.1 standard  
for the emerging Voice over IP market.  
AudioCodes' product line includes enabling  
technology products such as Voice over Packet  
chip processors, VoIP communication boards,  
VoIP media gateway modules and CPE devices.  
In addition, AudioCodes provides OEMs with  
media gateway system solutions for packet  
networks in the wireline, wireless, broadband  
access and media server markets.  
Conferencing  
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Fax Messaging  
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Supports up to 96 ports - PCI bus or packet streaming based  
T.37 store and forward  
Fax Relay  
Real-time fax over IP/T.38 compliant, automatic fallback to G.711  
Host-based Architecture - Media Stream over PCI  
Distributed Architecture - Media Stream over VoIP RTP  
Supports up to 480 ports of:  
ASR - 3rd party  
Recognition Engines  
Voice Processing  
International Headquarters  
4 HaHoresh Street  
Yehud, Israel 56470  
Tel: +972-3-539-4000  
Fax: +972-3-539-4040  
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G.711, G.723.1, G.729A, G.726/G.727, NetCoder  
Additional coders supported - contact AudioCodes for further information  
Voice Activity Detection (VAD) and CNG  
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US Headquarters  
2890 Zanker Road, Suite 200  
San Jose, CA 95134  
Tel: +1-408-577-0488  
Fax: +1-408-577-0492  
Echo Cancellation: G.168 with tail of 30 msec, 64 msec and 128 msec  
Trans-coding of G.711 RTP to any Low Bit Rate Coder RTP stream  
Gain Control: Automatic (AGC) or Programmable  
Packet side or PSTN side, DTMF and tone detection and generation, RFC 2833  
AudioCodes' proprietary TPNCP, MGCP (RFC 2705), MEGACO (H.248), SIP  
SNMP V2: Standard MIB-2, RTP MIB, Trunk MIB, AudioCodes' proprietary MIB  
Embedded Web Server  
In-band/Out-band Signaling  
Control  
Management Interfaces  
China  
Room 209, Tower A, Grand Pacific Building  
8A Guanghua Road  
Operating System  
Signaling  
7 Windows NT, 2000, XP 7 Linux 7 Solaris on Intel/Sparc  
Chao Yang District  
Beijing 100026, P. R. China  
Tel: +86-10-6581-6622  
Fax: +86-10-6581-5599  
PSTN  
CAS T1 robbed bit, MFC/R2 numerous country variants  
CCS ISDN PRI: numerous country variants including ETSI  
EURO ISDN, ANSI NI2, DMS, 5ESS, Japan INS1500  
IUA over SCTP per RFC 3057/2960  
Japan  
S Building B1F, 2-30-2  
Chuou, Nakano-ku  
Tokyo 164-0011, Japan  
Tel/Fax: +81-3-5348-1405  
SIGTRAN  
SS7 MTP2 link termination M2UA and M3UA over SCTP  
Hardware Specifications  
Ethernet  
Dual redundant 100 Base-T ports  
E-Mail info@audiocodes.com  
Hot Swap  
Physical Interfaces  
Full hot swap supported per PICMG 2.1  
Form factor - 6U PICMG 2.0 single cPCI slot  
TDM Interface - H.110 CT Bus  
www.audiocodes.com  
© 2003 AudioCodes Ltd. All rights reserved. AC, AudioCodes, AudioCodes  
logo, AudioCodes, IPmedia, Mediant, MediaPack, MP-MLQ, NetCoder, Stretto,  
TrunkPack and VoicePacketizer are trademarks or registered trademarks  
of AudioCodes Ltd. All other marks are the property of their respective  
owners. The information and specifications in this document and the  
product(s) are subject to change without notice.  
Telephony - two 50-pin Telco connectors on rear panel  
Ethernet - cPSB PICMG 2.16 on the backplane, Dual RJ-45 on rear panel  
40.7W, 3A at 5V, 7.8A at 3.3V  
Power  
Ref # LTRT-00244 01/03 V.3  
1 Integrated Partner Technologies  
2 Available in Q1/03  
3 Future release, may affect density  

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