Chapter 1 - Cisco Voice Gateways

• Analogue Circuits – Amplitude, Frequency,
Wavelength
• Digital Signalling – What is the difference between TDM and VoIP? – They are both digital signalling!
• ISDN – Also digital.
• Call control signalling in TDM – CAS, CCS, R2,
SS7.
• E & M Signalling

• ANI - Automatic Number Identification (ANI) refers to the telephone number of the calling party.
• CAC - Call Admission Control (CAC) is the tracking of bandwidth resources and traffic requests for the purposes of preventing oversubscription within the constraints of the current bandwidth. CAC prevents endpoints from sending more information, or attempting to send more information, than the available bandwidth can handle.

• Caller ID - Caller ID (CID) describes the delivery of a calling name or number, or both.
• CLID - Calling Line Identifier (CLID) is similar to ANI and refers to the number of the calling party.
• CNID - The Calling Number Identification or Calling Name
Identification is the same as the ANI.
• DNIS - The Dialled Number Identification Service is the telephone number of the called party. This refers to the number that the telephone company sends to the destination.
The DNIS may or may not be the actual number the calling party dials to reach the destination. For example, the caller may dial one number, but it may be translated or redirected by the telephone provider and those digits would be sent to the destination.

• Serialization -is the insertion of bits onto a link.
Serialization delay is the amount of time it takes for a networking device, such as a router, to encode a packet onto the wire for transmission. Serialization delay is incurred when encapsulating or segmenting a data stream into packets for egress from a given interface. The interface must service the packets one at a time, which in turn results in the delay.
• VAD - Voice Activity Detection is a voice encoding algorithm that takes note of silence during voice conversations and suppresses the transmission of voice packets that contain no actual data within them.

• Interfaces IP and PSTN networks
• Performs call setup and teardown between IP and PSTN
• Relays DTMF tones
• Supports IP and TDM control protocols
• Supports analogue FAX machines

• Foreign Exchange Office (FXO)
• Ear and Mouth (E&M)
• Foreign Exchange Station (FXS)
• E1 Primary Rate Interface ISDN
• Basic Rate Interface ISDN
• E1 R2 – (Serial cable) – Leased line
R1# show version (will show all interfaces)
R1# show voice port summary

• Cisco IOS Routers - 1700, 2600, 2800, etc
• Standalone Voice Gateways – Cisco Analogue
Telephone Adapter (ATA) devices, and Cisco
Voice Gateway (VG 200 Series)
• Switch Modules - Cisco Catalyst 6500 switches and 7600 Series routers

• When choosing a voice gateway for Cisco VoIP networks, it is important to ensure that the selected gateway supports the following four core requirements:
• 1. Dual Tone Multi Frequency Relay
2. Supplementary Services
3. Communications Manager Redundancy
4. Call Survivability

• Cisco voice gateway solutions are divided into three distinct categories. These categories are:
• 1. Cisco IOS Routers
2. Standalone Voice Gateways
3. Switch Modules

• The DSP is used in the voice gateway router to convert analogue voice signals to digital signals suitable for the IP network inside.
• DSP’s provide 4 critical functions in gateways;
• Voice Termination
• Transcoding
• Conferencing
• Media Termination Point
• http://www.youtube.com/watch?v=6hfkJPwP
5X4

• Voice termination applies to a call that has two call legs, one leg on a time-division multiplexing (TDM) interface and the second leg on a Voice over IP (VoIP) connection.
• This termination function is performed by digital signal processor (DSP) resources.

• After a WAN-enabled network is implemented, voice compression between sites represents the recommended design choice to save WAN bandwidth.
• This choice presents the question of how WAN users IP-enabled applications, which support only G.711 voice connections.
• Using hardware-based transcoding services to convert the compressed voice streams into
G.711 provides the solution.

• Connecting sites across an IP WAN for conference calls presents a complex scenario.
• In this scenario, the modules must perform the conferencing service as well as the IP-to-IP transcoding service to uncompress the WAN IP voice connection.
• In the Figure a remote user joins a conference call at the central location. This three-participant conference call uses seven DSP channels on the
Catalyst 4000 module and three DSP channels on the
Cisco Catalyst 6000.

• This three-participant conference call uses seven DSP channels on the Catalyst 4000 module and three DSP channels on the
Cisco Catalyst 6000.
• The following list gives the channel usage:

Cisco Catalyst 4000
• – One DSP channel to convert the IP WAN G.729a voice call into G.711
• –Three conferencing DSP channels to convert the
G.711 streams into TDM for the summing DSP
• –Three channels from the summing DSP to mix the three callers togethe r
Cisco Catalyst 6000
• – Three conferencing DSP channels. all voice streams get sent to single logical conferencing port where all transcoding and summing takes place.

• A Media Termination Point software device allows Cisco Unified Communications
Manager to relay calls that are routed through
SIP or H.323 endpoints or gateways.
• You can allocate a media termination point device because of DTMF or RSVP requirements. When a media termination point is allocated for RSVP, you can insert it between any type of endpoint device, including SIP or H.323 devices.

• Voice Gateways
• Circuit signalling
• Analogue Circuits
• FXO & FXS Ports, E&M ports
• Digital circuits – TDM
• ISDN
• DSP Resources including Voice Encoding