The Voice T1 and Services Module establishes the Cisco Catalyst® 6000 family as the most complete campus multiservice platform available, providing high-end T1 or E1 gateways to the Public Switched Telephone Network (PSTN) or legacy private branch exchanges (PBXs) and network-based voice services.
The Catalyst 6000 family brings data, voice, and video integration onto the campus for fully integrated communications on every desktop. Campus multiservice networking, or convergence, provides voice support using the IP network infrastructure rather than the traditional PBX. This drastically increases the leverage of telephony spending into overall infrastructure spending, reduces capital and operational costs, and opens the environment to innovation by supporting new applications. (See Figure 1.)
Cisco is using the Catalyst 6000 family to lead its customers to campus convergence. The Voice T1 and Services Module, also available in an E1 variant, introduces the following new features for campus multiservice networking:
Campus Multiservice Networking
- Common channel signaling (CCS)In this mode there are 23 DS0 channels for T1 and 29 for E1 for voice traffic; the 24th T1 DS0 or 16th E1 channel is for signaling. Any channel can be configured for common channel signaling.
- Integrated Services Digital Network (ISDN) Primary Rate Interface (PRI) signalingEach interface supports 23 channels for T1 and 30 channels for E1. The default mode is for the 24th T1 channel or 16th E1 channel to be reserved for signaling. Both network side and user side operation modes are supported.
The Voice T1 and Services Module provides transcoding and conferencing services for the multiservice network. Transcoding enables a full voice-compression solution by offering transcoding services to endpoints not capable of supporting compressed voice or a different encoding type to the remote end. Figure 2 shows an example of transcoding.
Figure 2 Example of Transcoding
In the top part of Figure 2, a call is completed between two endpoints capable of supporting compressed voice. The bottom of Figure 2 shows what happens when the call is transferred to an endpoint, such as voice mail, that does not support compressed voice.
Figure 3 shows an example of the Voice T1 and Services Module in a conference bridge application. In Figure 3, IP Phone B conferences IP Phones A and C. The Cisco CallManager directs the media stream to the Voice T1 and Services Module, which bridges the media streams together.
Figure 3 Voice T1 and Services Module Used in Conference Bridge Application
- T1 CCS signaling
- T1 binary 8-zero substitution/alternate mark inversion (B8ZS/AMI) line coding, u-law or a-law coding
- E1 CCS signaling
- E1 HDB3 line coding
- ISDN PRI, network and user side operation modes
- Echo cancellation
- Silence suppression, Voice Activity Detection (VAD)
- Comfort noise generation
- G.711 to G.723 and G.729a transcoding (max. 8 x 32 channels of transcoding)
- Conference bridging, meet-me and ad-hoc conference modes (max. 8 x 16 channels of conferencing)
- Fax pass-through
- Network management support
- Simple Network Management Protocol (SNMP) compliant
- CiscoWorks interface for configuration
- Cisco Voice Manager (CVM)
- Cisco CallManager
- Switched Port Analyzer (SPAN) or port mirroring support
- Dial control Management Information Base (MIB) (RFC 2128) and CISCO dial control MIB ext. to RFC 2128
- CISCO-VOICE-DIAL-CONTROL-MIB Voice Dial Control MIB
- CISCO-VOICE_IF_MIB Voice Interface MIB
- CISCO-VOICE-ANALOG-IF_MIB Voice Analog Interface MIB
- CISCO-DSP-MGMT-MIB Digital Signal Processing Management MIB
- RFC 1157 SNMP
- RFC 1643 Ethernet MIB
- RFC 1493 Bridge MIB
- RFC 1213 MIB II
- RFC 1573 MIB II Interface Extensions
- RFC 1757 Ethernet RMON
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