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Optical STM-1 over Gigabit Ethernet 63E1 to SIP converter
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Product Details
| Brand Name | baudcom | Place of Origin | China |
Product Description
Optical STM-1 over Gigabit Ethernet 63 E1 to SIP converter is a carrier grade VoIP gateway, which is designed for telecom operators, ISPs with high reliability and performance. Focusing on a concept of maintainable, manageable and operable, 63E1 to SIP converter adopts STM-1 interface which features high integration and large capacity. It provides carrier-grade VoIP and FoIP services, as well as value-added functions such as modem and voice recognition. Thus it constructs a flexible, high-efficient, future-oriented communication network for users.
The VOIP media gateway supports a range of signaling protocols, realizing the interconnection between SIP and traditional signals like SS7 and PRI. The 63 E1 to SIP converter supports multiple codec methods, offers signal encryption technology and smart voice recognition technology, and improves the utilizing efficiency of trucking resources while ensuring voice quality. The trunk gateway is ideally fit for various networks of ISPs, telecom operators and large-scale enterprises.
Key FeaturesCarrier grade hardware design, 1+1 power supply and MCU, hot plug
High-integrated structure, STM-1 155M (63*E1) in 2U size
Support flexible dialing rules and operations, allowing users to customize dialing rules according to different working environments
Support multiple coding standards: G.711A/U, G.723.1, G.729A/B and iLBC
High compatibility, interoperable with PBX of Avaya, NEC and Alcatel, and also leading soft-switch of Huawei, Cisco and ZTE etc.
63E1 to SIP Converter ApplicationSDH Interfaces 2*Standard LC SDH, 155M
1+1 Redundancy Channels Protection
Master/Slave Clock SourceMain Control Unit(MCU) 1+1 Redundancy, Hot Plug DigitalProcessing Unit (DTU) 4*DTU Maximum
Support 512 Voice Channels Each BoardEthernet Interface GE1: 10/100/1000 BaseT Adaptive Ethernet
GE0: 10/100/1000 BaseT Adaptive EthernetSerial Port 1* RS232, 115200bps Call Features
Flexible Route MethodsPSTN-PSTN, PSTN-IP，IP-PSTN
Intelligent Routing Rules
Call Routing base on Time
Call Routing base on Caller/Called Prefixes
256 Route Rules for each Direction
Caller and Called Number ManipulationVoice Capabilities Codecs:G.711a/μ law,G.723.1, G.729A/B, iLBC, AMR
Silence Suppression
Comfort Noise
Voice Activity Detection
Echo Cancellation (G.168),with up to 128ms
Adaptive Dynamic Buffer
Voice, Fax Gain Control
FAX: T.38 and Pass-through
Support Modem/POS
DTMF Mode: RFC2833/Signal/Inband
Clear Channel/Clear ModeSoftware Features Local/Transparent Ring Back Tone
Overlapping Dialing
Dialing Rules，with up to 2000
PSTN group by E1 port or E1 Timeslot
IP Trunk Group Configuration
Voice Codecs Group
Caller and Called Number White Lists
Caller and Called Number Black Lists
Access Rule Lists
IP Trunk PriorityEnvironmental Redundant Power Supply
Power Supply: 100-240VAC,50-60 Hz
Power Consumption:110W
Operating Temperature:0 ℃ ~ 45 ℃
StorageTemperature: -20 ℃ ~80 ℃
Humidity:10%-90% Non-Condensing
Dimensions(W/D/H):437*320*88mm(2U)
Unit Weight: 6.5kg
Compliance: CE, FCC,CCCPSTN
ISDN PRI:23B+D(T1),30B+D(E1),NT or TE
ITU-T Q.921, ITU-T Q.931, Q.SigSignal 7/SS7: ITU-T, ANSI，ITU-CHINA
MTP1/MTP2/MTP3, TUP/ISUPE1 Frame Type: DF,MF_CRC,MF Line Code: HDB3 Clock Source: Local/Remote Clock Source Maintenance Web GUI Configuration
Data Backup/Restore
PSTN Call Statistics
SIP Trunk Call Statistics
Firmware Upgrade via TFTP/FTP/Web
Network Capture
SNMP v2
Syslog: Debug, Info, Error, Warning , Notice
Call History Records via Syslog
NTP Synchronization
Centralized Management System
Contact Us
- Shanghai Baudcom Communication Device Co.,Ltd
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| electronic communication equipment | phone |
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