Four-Port Voice Interface Card Module - FXS and DID (VIC3-4FXS / DID =)

Cisco siping / fax network modules for the Cisco 2800, 2900, 3800, and 3900 Series Unified Communications Routers provide support for packet voice technologies, including VoIP (H.323, Media Gateway Control Protocol [MGCP] and Session Initiation Protocol [SIP]). Cisco Unified Communications Solutions provide the means to integrate voice and data within a single network, allowing users to take full advantage of services such as IP telephony, integrated services, and payment bypass, while simultaneously increasing productivity. Managed by Cisco IOS software, these solutions include enhanced Quality of Service (QoS) features, intelligent network queuing, and standardized encapsulation, providing efficient simultaneous transmission of voice and data directly over IP networks. Cisco IOS software solutions allow you to transmit voice traffic that is time-sensitive, even over low-bandwidth WAN connections with the necessary priority and voice and fax quality. Voice over IP networks provide data transfer flexibility with routing capabilities (dedicated lines, Frame Relay and ATM).

Cisco Voice / Fax Network Modules provide Cisco Unified Communications Gateway for PSTN calls and legacy telephone equipment. Users can deploy networks that take full advantage of investing in existing telephone equipment, as well as deploy or integrate IP telephony at this time or in the future. These network modules enable users to work anywhere in the infrastructure of integrated voice, video and data transmission, as well as add connections to traditional telephony and IP telephony channels.

Cisco VIC3-4FXS / DID Specifications
physical characteristics
Interface Type: VIC
Status Indicators: LED Indicators for Port Status and Voice Processing Resources
Interface ports
Interface ports: 4 x RJ-11 FXS / DID channels
Data transfer rate:
IP Telephony
Unified Communications: • Unify all Cisco IP communications solutions by providing flexible and reliable connections to public or private telephone networks around the world.
• Provides a gateway for connecting Cisco IP Phones to PSTN or traditional PBX or PBX
• Provides a gateway for connecting traditional PBXs, telephones, fax machines to PSTN and push-button communication systems to voice, data and video infrastructure
Bypass payment: • Reduce or eliminate payment for the use of long-distance and local communications by transmitting voice and fax traffic through a corporate network, LAN, urban network (MAN) or WAN.
• Works with existing telephones, fax machines, PSTN and push-button systems.
• Interconnecting channels completely replace the structure of connecting lines (digital-digital, digital-analog or analog-analog)
• Compatible with all Cisco IP IP phones, analog phones, fax machines, PBX or PBX connections with other Cisco products that support voice transmissions
Voice transmission via packet transport: • Voice and fax data over IP-VoIP traffic at level 3 can be transmitted through any 1 or 2 level carrier, including ISDN, leased lines, serial connections and Ethernet.
• Compressed Real-Time Protocol (cRTP) offers RTP header compression and packet fragmentation methods that provide high-quality voice and fax data transmission over any WAN connection
• Call Admission Control and PSTN reservations use the Service Assurance Agent (SAA) to determine delay and jitter, provide Calculated Planning Impairment Factor (ICPIF) real-time calculations before setting up a call through an IP infrastructure. SAA packages emulate voice packets and get the same network priority.
• Enhanced QoS mechanisms: these configurable features of Cisco IOS software reserve the necessary bandwidth, prioritize voice and fax traffic to ensure transparent transmission of high-quality voice and fax data. These include Resource Reservation Protocol (RSVP), queuing methods (such as Low Latency Queuing), IP priority, and differentiated services code point (DSCPs)
Signaling call control: Supports H.323 V1 / V2 / V3 / V4, MGCP 0.1 / 1.0, and SIP call control protocols. Also supports Cisco UCM using MGCP, H.323 or SIP
The standard voice codecs of the International Telecommunications Union (ITU): G.711, G.729, G.729a / b, G.723.1, G.726, G.728, iLBC, G.722 — these standardized compression methods provide voice over IP.
Signaling support on telephone interfaces: Supports the following signaling protocols:
• FXO / FXS loop-start and ground-start signaling
• Inbound signaling (such as dual-tone multifrequency [DTMF], multifrequency support)
• Country-specific signaling
Voice features: • Echo cancellation at the tail of the signal up to 32 ms (adjustable tail length)
• Silence suppression, Voice Activity Detection (VAD) - bandwidth is used only during a conversation. During silence during a telephone conversation, bandwidth is used to transfer data.
• Generation of comfortable noise
• Private Line Automatic Ring-Down (PLAR) —Provides a direct connection to another digital or analog voice port when you pick up the handset. Includes "Trader Turret" PLAR
• Local / Advanced Voice Busy-Out-Automatically switches any desired voice channel to a PBX or PSTN when a direct WAN or LAN connection fails to the router or any part of the network.
• Caller ID support — Caller ID configuration on each port (with call unlocking) on analog FXS, FXO, and DID interfaces
• Interception groups between cards - calls can be redirected to the first available line
• Built-in add and remove multiplexer (delete and insert) - adds and removes multiplexing for voice in a two-port network module. Eliminates the need, support and cost of using external multiplexer addition and removal
• Linking dial plans — simplifying configuration and management by automatically linking dialed phone numbers and IP addresses
• Interactive Voice Response (IVR) support — Provides the Attendant, voice mail support and call routing.
• Hoot and Holler over IP - Provides high-quality multicast voice services Hoot and Holler and multicast conferencing over existing WAN endpoints
Features of voice ports: • FXS - provides detection of reversal of battery polarity, initiation of monitoring and response monitoring off
• Disabling any port without negative impact on other ports of the same module.
Fax and modem: • Fax and modem traffic passes through the voice port
• Fax Switching - provides a more reliable protocol for sending faxes on packet networks. T.37 and T.38 fax protocols are also supported.
High-Performance Flexible Digital Signal Processor (DSP) architecture: • Channel Capacity - Supports up to 48 voice channels
• Flexible DSP architecture - no need to specify the complexity of the codec in the configuration. The appropriate codec is automatically selected when the connection is established and the DSPA resources are automatically distributed.
• Feature Updates — DSP architecture allows for the addition of new features through simple code updates.
Network Management Support: • Cisco UCM
• Simple Network Management Protocol (SNMP) Compatibility
• Management with Management Information Base (MIB)
• CiscoView interface for configuration
• ConfigMaker
• NetSys support
Functions: • Signaling Addressing Formats:
- In-band DTMF
- Out-of-band ripple (8-12 packets per second)
• Signal transmission modes:
- FXS: loop-start, ground-start
- DID: Immediate, delay dial, wink start
• DID loop resistance: up to 1800 ohms (including terminal equipment)
• FXS loop resistance: up to 600 ohms (including telephone and terminal equipment)
• Voltage with the tube lifted: - 44 V
• Current with the tube down: 25 mA (maximum)
• Ringing tones: customized for requirements of different countries
• Call voltage: 45 Vrms in 5 REN with zero loop length (balanced)
• Call frequencies: 20 Hz, 25 Hz, 30 Hz, 50 Hz
• REN load: 5 REN per port (maximum), 8 REN per VIC (maximum) and 2 REN per port in FXS / DID mixed mode
• Shutdown control: power failure (group call control, remote shutdown)
• Caller ID: transmitting frequency shift keying (FSK) data while off-hook
• Loop length: up to 450 m of twisted pair cable category 5, 24 AWG