Network and VoIP office gateway IP-M

ROUTER AND VOIP GATEWAY

Technical Characteristics

Protocol supported STP, IP, ICMP, TCP, UDP, ARP, FTP, TFTP, DNS, HTTP, SNMP, MIB II, NAT, PAT, DHCP, VLAN (802.1q), PPPoE, SIP
Functional Router / Bridge mode, IP-Filtering, Smart Ethernet Switch , 24 Line PABX, SIP v2.0 Gateway
Connections 6 10/100Base-TX 2 100Base-FX, 32 FXO/FXS (optional), 1 E1 (R1.5, EDSS) (optional)
Managment WEB, SNMP, SNMP Trap
Speech coding G.711, ACELP 4,8, G.729ab
Statistics By every Ethernet Port. Calls of PABX and SIP gateway.
Dimensions 380x260x105 (19" rack 1U) ~7Kg
Power supply 110/220V ~50Hz

As a switch, LAN:

  • Implementation of STP protocol
  • Collection of statistics (onsent/received Ethernet frames) via each port and its transmission to external equipment
  • Control the status of Ethernet connections and automaticaly send the SNMP message on connection / disconnection to external equipment
  • Control of parameters via SNMP (version 2) protocol: configuration and state of ports, port priority command on/off, establishment of priority levels via ToS, port based VLAN mode on/off, input and output flow rate of each port.

As an IP-router:

  • Filtration of IP packets via SNMP (version 2) protocol to the complete criteria set
  • Translation of network addresses according to the NAT (network address translation) and PAT (port address translation) technology
  • Support of traffic priorities: via ToS fields, via IP-address of receiver and sender
  • Support of dynamic routing
  • Collection and transmission of statistics via IP, TCP, UDP, ICMP protocols and IP-network usersџ Control of parameters via assigned interfaces: filtering, network address translation, priorities and other functions.

As an IP-PBX:

  • Establishment of connections between internal FXS users
  • Establishment of connections via SIP protocol
  • Support of modes of proxy server and SIP customer
  • Connection of payphones via FXS interface and provision of connection with payphone management center via IP channel
  • Simultaneous implementation of up to 12 connections with 8 Kbit/s codec, 4.8 Kbit/s ACELP to operate on low-speed IP-channel
  • Control of equipment parameters and line features, user rights, SIP protocol settings, speech coding mode via SNMP (version 2) protocol.

As a E1 gateway:

  • Transformation of speech and service information, received from PBX in terms of bit transfer rate, into IPflow and reverse transformation from IP-flow into bit transfer rate
  • Support of protocols of telephone signaling: R 1.5 and EDSS1
  • Support of protocol of SIP 2.0 IPtelephony network signaling
  • Two-way conversion of telephone signaling into IP-telephony network signaling
  • Coding of speech in synchronic mode according to the G.711 recommendation
  • Coding of speech in packet mode according to the G.711, ACELP 4.8, G.729ab recommendation

Documentation

CT-IPM VoIP gateway 4,7Mb