Voip

VoIP Telephone Initialization Process

1. VoIP phone receives PoE 2. Network switch id's VoIP phone with VLAN 3. VoIP phone requests IP address from DHCP 4. VoIP phone contacts TFTP 5. VoIP phone registers with CME using Skinny (SCCP) and is ready to place/receive calls

Purpose of Call Manager Express (CME)

A VoIP management application which runs on an Integrated Services Router's (ISR) IOS in order to provide call processing in a VoIP network.

Skinny Station Control Protocol (SCCP)

A cisco proprietary protocol designed to be utilized between the call manager device (CM or CME) and the subscriber terminals (phones). Allows you to quickly make changes to features and funtionality on a Cisco VoIP Network.

Media Gateway Control Protocol (MGCP)

A client-server call control protocol used by call-control devices (Call Manager) to manage and contol IP gateways.

Purpose of Dial Peers

A software structure that binds a dialed digit string (phone number) to a voice port or IP address of the destination network.

H.323 protocol

A suite of peer-to-peer protocols (e.g. call signaling, audio/video codecs, etc) that provide IP networks with traditional telephony functionality.

Voice Over Internet Protocol (VoIP)

A technology which allows Internet Protocol-based networks to be utilized for real time voice applications.

Quality of Service (QoS)

Ability of the network to provide better or special services to a set of users and applications at the expense of other users and applications.

Purpose of the Real Time Monitoring Tool (RTMT)

Allows administrators to collect, view, interpret and monitor the various counters, trace files, and logs generated by Call Manager.

Session Initiation Protocol (SIP)

An alternative to H.323, an ASCII-based peer-to-peer protocol which defines end to end call signaling between devices by using a text based HTTP-like model of communications.

Purpose of a Cisco Unified Communications Manager (CUCM) aka Call Manager

An application that runs on a dedicated server providing call processing services for up to 10,000 users.

Real-time Transport Protocol (RTP)

Based off of the User Datagram Protocol (UDP) which was created to provide end to end transmission of streaming media traffic. Used to communicate directly between the voice endpoints themselves.

2 types of call processing used within a VoIP network

Centralized Distributed

Components of QoS

Classification and Marking Congestion Management (Queing)

Purpose of an Inter-Cluster Trunk

Communications between clusters.

Benefits of VoIP

Cost Savings Bandwidth Rich applications and Features

Purpose of Signaling protocols

Exist to setup and maintain a bi-directional real-time media stream between VoIP subscriber terminals. Also, provides necessary signaling for VoIP networks to interface with PTSN

Purpose of Survivable Remote Site Telephony (SRST)

Fault Tolerance!! Within a Call Manager VoIP network. Can be enabled throughout a VoIP network to allow calls to be processed and routed between sites in the even that connectivity to the Call Manager is lost.

4 basic components of a VoIP Network

IP network Call processor (controllers) Media/Signaling gateways Subscriber terminals (endpoints)

Characteristics of the Voice VLAN

Increased performance Improved Manageability Physical topology independence Increased security

Cluster

Mulitple servers (max of 8) clustered together and managed as a single entity; up to 30,000 users per cluster.

Power options associated with Cisco IP phones

Power over Ethernet (PoE) - 15.4 Watts Power bricks (Powercords)

Call Admission Control (CAC)

Prevents IP calls from being extended across the WAN link, if the additional bandwidth required would exceed the CAC-allocated bandwidth for concurrent calls, thus preventing oversubscription.

Purpose of the Disaster Recovery System (DRS)

Provides a backup and restore capability written to a SFTP server.

Purpose of a Cisco Router Voice Port (Voice WAN Interface Card -VWIC)

Provides a physical interface on a voice-enabled router for a PRI T-1 to be connected from the CSN to the PSN.

4 main components of call routing on a Call Manager (CUCM) Network

Route patterns Route lists Route groups Gateways/Trunks

Role of the TFTP server

Serves as a central repository for all required phone configurations in the network.

Class of Control

The ability to apply calling restrictions to devices.

Convergence

The combination of voice, video, and data services across the same network infrastructure.

Transcoding

The direct digital-to-digital conversion from one codec to another.

Function of a Publisher Server

The first server in the cluster. The only server whose database can be altered.

Function of a Subscriber Server

The primary call controllers responsible for registering phones and setting up calls.

Functions of a call processor

call processing device control directory service computer telephony integration (CTI) trunking gateway support

Basic Phone-to-Phone Call Process w/in Call Manager

1. Caller dials distant end 2. Digits are received by Call Manager 3. CM looks-up the distand end phone's IP address 4. CM checks status of distant end phone 5. If ready to receive calls, CM rings the phone 6. Distant end phone hears ringing and picks up 7. CM connects the phones using RTP. 8. CM steps aside and phones communicate with one-another