Useful Information 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
Voice Terminology 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.
Voice Gateway 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
Voice Gateway Components 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
Voice Gateway Capabilities 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
Voice Gateway Types Cisco voice gateway solutions are divided into three distinct categories. These categories are: 1. Cisco IOS Routers 2. Standalone Voice Gateways 3. Switch Modules
Digital Signal Processor (DSP) 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 5X4 5X4
1- Voice Termination 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.
2 - Transcoding 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.
3 - Conferencing 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.
Conferencing - ctd This three-participant conference call uses seven DSP channels on the Catalyst 4000 module and three DSP channels on the Cisco Catalyst 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.
4 – Media Termination Point 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.
Chapter Summary – What you should study! Voice Gateways Circuit signalling Analogue Circuits FXO & FXS Ports, E&M ports Digital circuits – TDM ISDN DSP Resources including Voice Encoding