Datasheet

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Feature Description and Benefits
Voice over Packet Transport
• Voice/Fax over IP—VoIP traffic at Layer 3 can travel over any Layer 1 or Layer 2 media, including
ISDN, leased lines, serial connections, Frame Relay, Ethernet, Token Ring, and ATM.
• Voice/Fax over Frame Relay—Voice over Frame Relay is supported using FRF.11 and FRF.12
standards. This solution also uses features found only in Cisco IOS® Software for maintaining voice
quality.
• Voice over ATM is supported using AAL2 or AAL5 encapsulation. Uses existing ATM networks as a
direct transport method for voice. Requires ATM interfaces such as T1/E1 ATM, Inverse Multiplexing
over ATM (IMA), DS3/E3 or OC-3, or DSL WICs.
• Compressed Real-Time Protocol (cRTP) offers RTP header compression and packet fragmentation
techniques that allow toll-quality voice and fax transmissions over any WAN connection.
• Call Admission Control and PSTN Fallback uses Service Assurance Agent (SAA) to determine
latency, delay and jitter and provide real-time Calculated Planning Impairment Factor (ICPIF)
calculations before establishing a call across an IP infrastructure. SAA packets emulate voice packets
receiving the same priority as voice throughout the entire network.
• Advanced QoS Mechanisms—These configurable Cisco IOS Software features reserve appropriate
bandwidth and prioritize voice and fax traffic to help ensure transparent delivery of toll-quality voice
and fax. They include Resource Reservation Protocol (RSVP), queuing techniques (such as Low
Latency Queuing), IP Precedence, and differentiated services code points (DSCPs).
Call Control Signaling
Supports H.323 V1/V2/V3/V4, MGCP 0.1/1.0, and SIP call control protocols. Also supports Cisco
CallManager using MGCP or H.323.
International Telecommunications
Union (ITU) Standard Voice Codecs
G.711, G.729, G.729a/b, G.723.1, G.726, G.728, and GSM—These are standards-based compression
technologies allowing transmission of voice across IP, Frame Relay, and ATM. The G.711 standard
employs 64 kbps PCM modulation using either u-law or A-law. Other codecs employ lower bit rates.
Telephony Interface
Signaling Support
Supports the following signaling protocols:
• FXO/FXS loop-start and ground-start signaling
• E&M (wink, immediate, delay)
• Inbound signaling (such as dual-tone multifrequency [DTMF], multifrequency support)
• T1 and E1 channel associated signaling (CAS)
• T1 and E1 PRI Q.931 user side and network side
• T1 and E1 PRI QSIG
• E1 MelCAS
• E1 R2 CAS
• T1 and E1 Transparent common channel signaling (CCS) (with multiple-D channel)
• Country-specific signaling
Voice Features
• Echo Cancellation—Cancels echo on tail circuits up to 32 msec (configurable tail length)
• Silence suppression, voice activity detection (VAD)—Bandwidth is used only when someone is
speaking. During silent periods of a phone call, bandwidth is available for data traffic.
• Comfort Noise Generation—This feature reassures the phone user that the connection is being
maintained, even when no voice packets are being transmitted
• Private Line Automatic Ring-Down (PLAR)—Provides a direct connection to another digital or
analog voice port by lifting a telephone handset on one end. Includes “Trader Turret” PLAR
• Local/Advanced Voice Busy-Out—Automatically busies out any desired voice trunk line to a PBX or
PSTN when a direct WAN or LAN connection to the router or any part of the network to the destination
port is down