CCNA Voice Preparation

CCNA Voice Summary – Part 12

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Administrative User Interfaces for Unified Communicatin Products – CUCM, Unity Connection, CUPS, CUCX.

Unified OS Administration – for software management upgrades, tftp management; time, security – IPSEC tunnels, certificate management; interface specifics etc. All SIP phones get their times from the NTP and the SCCP from CUCM.
Unified Servicability – check which network services are running, start and stop them; application specific services; alarm configuration; trace and debug; snmp configuration.
CLI – command line interface
Unified Reporting– Runs reports about the system
Disaster Recovery System – Backup; Restore; Help

Cisco Unified CM Administration Interface

There are three Entities in UCM:
Something specific -> A group. A group can have several specifics -> A list. A list can have multiple groups.
A user -> A user is assigne Roles -> The roles depend on the group that they are in.
A user is assigned to a group. The group has roles and the roles define the permissions.

An application user – he does not appear in the directory such as the CCM System User,
An End user shows up in the corporate directory, Has a phone. LDAP only popullates the end user list. LDAP is a one way synchronization.

CCNA Voice Summary – Part 11

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Quality of Service

Disadvantages of packet switched networks can be solved using QoS.

 ITU recommendations (G.114 specifications):

Delay < or = 150 ms one way

Jitter < or = 30 ms

Packet loss < or = 1%

Bandwidth for Voice and Video RTP is subject to codec, sampling, L2 and L3 overhead. Bandwidth for signaling depends on the protocol.

Bandwidth for Video should always have 20% additional overhead for key frames where the entire picture is sent rather than the normal pixel difference.

 Types of QoS Models:

Best Effort – No QoS. Used FIFO – first in first out.

Integrated Services (IntServ) – Guaranteed bandwidth using RSVP. In order for it to work, it had to be enabled on every Layer 3 device in the path of a call.

Differentiated Services (DiffServ) –  Guaranteed bandwidth for voice and signaling. Classified different traffic types. Uses policies to ensure differentiated levels of priority and service. Most commonly sued in enterprise networks because carriers usually use MPLS-TE. Does not have to be enabled on every router in a VoIP call path. First we need to classify and mark the traffic. We then deal with queuing and dropping mechanisms. Voice should be queued first. Finally have link specific mechanisms like traffic shaping, fragmentation, compression then transmission.

 Where do we apply QoS:

Classification and Marking – as close to the source as possible.

Policing –   this is dropping the traffic. It is always applied at the ingress port.

Scheduling (Queuing, congestion management and congestion avoidance) – applied at the egress.

Link specific mechanism – always the last to be done as the packet is leaving the device.

 QoS Queuing Types:

FIFO –  First In, First Out

Priority Queuing (PQ) – sends all the packets in a PQ then sends from other Q’s. We first service the priority queue then service the rest of the queues. This can lead to starvation of other queues especially when the PQ is always full.

Weighted Fair Queuing (WFQ) – Schedule traffic to be transmitted based on flow.

Class- Based Weighted Fair Queuing (CBWFQ) –  Schedule traffic to be transmitted based on admin defined classes to define which traffic gets how much of the available bandwidth.

Low Latency Queuing – Basically a combination of PQ and CBWFQ.

 Link Specific Mechanisms : Fragment and interleave (LFI)

Serialization delay – time required to serialize the frames and put them on the wire. For slow links, this delay is a major factor affecting latency and jitter. For slow links, data packets which are large need to be fragmented and interleaved with the smaller urgent voice packets.

As we go above a link speed of 768 kbps, LFI and compression end up causing a negative value in terms of CPU etc.

Voice packets should never be fragmented because it will introduce additional delay.

 Link Specific Mechanisms: Header and Payload Compression

IP /UDP / RTP headers can be reduced from 40 bytes to 2 bytes with a standard UDP or 4 bytes if UDP Check summing is used. This is only for low speed links , or = 768 kbps.

Payload compression – compresses not only the RTP header but also the RTP payload. It is extremely CPU intensive.

 Link Specific Mechanisms: Traffic Shaping

Policers drop the traffic and a given line rate. Traffic shaper delays or queues the excess traffic over the line rate and releases the traffic over a given time rate. This smooths out the traffic, prevents unnecessary drops and uses the available bandwidth more efficiently.





CCNA Voice Summary – Part 10

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 Cisco unified Presence server CUPS

It is a presence add-on to CUCM. It provides status information and enterprise instant messaging. Integration with CUPC -Cisco Unified Personal Communicator, IP Phone Massager IPPM (IM on a phone), Instant Messaging IM, 3rd party presence integration through SOAP and AXL. Uses standard SIP to collect data. It integrates into nearly every IT stuff (CUCM, IP Phones, Unity, LDAP server etc.). We have the ability to cluster. Up to 15000 users per cluster.

Ability to provide Inter-domain federation with other presence vendors via SIP,  extensible Messaging and Presence Protocol XMPP or Jabber Extensible Communication Platform XCP e.g. Other CUPS clusters, WebEx Connect, Microsoft Lync, Google Talk

 CUPS Features:

Built upon cisco client Services Framework CSF. Standalone softphone or desktop control – fully featured or where CUPC softphone controls the desk phone. Softphone mode supports SRST failover.IM Support. Voice and Video Chat. Conferencing via cisco meeting place or cisco WebEx. IMAP integration with Unity /Unity Connection for MWI. LDAP integration for directory services.

 CUPS Traffic Flow:

SIP/SCCP Signaling between endpoints and CUCM. RTP media between endpoints (CUPC AND phone).

CUPC and CUCM if using softphone mode – sip/ simple

CUPC and CUCM if using desktop mode –

CUCM and CUPS – CTI (JTAPI), DB replication to CUPS, SIP/SIMPLE.

CUPC and CUPs – 7.x clients use sip only. 8.0 clients use XMPP protocol.


CCNA Voice Summary – Part 9

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Cisco Voicemail Options -Cisco Unity Express, Cisco Unity Connection and Cisco Unity

Cisco Unity – Original release. It took very long to set up. Runs on windows and fully integrates into Exchange and Lotus Domino. This is fading away. it is being replaced by unity connection. Its greatest advantage was the fact that it integrates directly into exchange.

Cisco Unity Connection (CUCx)

Linux based appliance. It provides integrated messaging server. It was an IMAP only integration but it recently integrates with MS Exchange. Used for the larger deployments up to 20,000 users per server. Can cluster in an active deployment. Fully features auto attendant and voice menu server. Directory services synchronization with TTS text to Speech and ability to synch from LDAP. MS for both calendaring and IMAP. It can be used for SRSV survivable Remote Site VoiceMail for fallback. Multiple CUCx clusters form relationships using Digital networking. If it’s Unity Connection to Unity Express or third party communication, then we use a standard known as VPIM Voice Profile Internet Mail for networking to other VM servers. VPIM uses SMTP messages. Messages are stored in the local CUCx file system. Can support fax messaging.

CUCx Clustering:

Max 2 servers config that can be active active. Both servers process RTP, HTTP, and IMAP. Uses the same Informix DB structure as CUCM. Pub server owns the database and message store. Sub server subscribes to both.

CUCx Traffic Flow:

SIP/SCCP Signaling between endpoints and CUCM. SIP only or SCCP only signaling between CUCx and CUCM. RTP media between the CUCx and the endpoints (phones and gateway.

 Cisco Unity Express CUE

Linux OS IOS like CLI. Integrates with CME. Uses the AIM (ISM – Internal service module) and NM(SM- service module) Module. AIM Module is a full blown computer. NM – full blown with bigger storage space, increased processor speed. Unity Express is designed for up to 250 users. Features: Basic IVR – Interactive Voice Response, Auto Attendant and Email Exchange Integration. Can support fax messaging. Can provide SRSV. Has fully featured IVR/ Auto Attendant/Voice Menu with the ability to queue callers when integrated with BACD.

CUE Traffic Flow:

SIP/SCCP Signaling between endpoints and CUCM. SIP only signaling between CUE and CUCM. RTP media between the CUE and the endpoints (phones and gateway.

CUE can be integrated directly with a larger CUCM server cluster – it will talk the JTAPI language.

 CUE ISR Specs:

ISR (28XX OR 38XX)


6 voice mail ports with 6 max concurrent communication

6 IVR sessions

65 voicemail boxes

14 hrs. VM storage


Default 8 voice mail ports to 24 ports with 24 max concurrent communications

24 IVR sessions

275 voicemail boxes

300 hrs. VM storage


ISR (29XX OR 39XX)

ISR-SRE-300 (integrated services module – service ready engine)

Default 2 voice mail ports to 10 ports with 10 max concurrent communications

10 IVR sessions

100 voicemail boxes

60 hrs. VM storage


Default 4 voice mail ports to 32 ports with 32 max concurrent communications

32 IVR sessions

300 voicemail boxes

600 hrs. VM storage









CCNA Voice Summary – Part 8

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 CUCM (Cisco Unified Communication Manager / Call Manager)

Fully featured Voice and Video.

Supports up to 30000 IP phones per cluster in reality around 20000. A cluster is a shared database. Multi-server redundancy. Multi-site support. Very expensive compared to the rest. It runs on a hardened red hat Enterprise Linux platform as an appliance with no direct shell can run as a IU rack mount or a UCS blade server running in VMware ESXi. It stores all the data in an IBM Informix Database.

 CUCM Features:

It is a call processing Engine

Configuration is done mainly via web UI

It can function as a music on hold and conferencing server.

Can provide directory services for users (integrated or LDAP sync e.g. Microsoft Active Directory) and Presence services.

It has its own integrated Disaster recovery Subsystem (DRS) for backup and restore.

Cisco Unified Serviceability and Cisco RTMT provide ability for management and troubleshooting of log files and alarms and real time information.

 CUCM Per Server Specs:

Cisco UCS Servers:

Cisco UCS C210 Rack Mount Server – 7500 endpoints / server

Cisco UCS B200 Blade server – 7500

Cisco UCS C210 Rack Mount Server – 1000

Min Requirements of any server:





MCS 7845-13 – 10000 ENDPOINTS

MCS 7845 – 7500

MCS 7835 – 2500

MCS 7825 – 1000

MCS 7816 – 500

MCS 7815 – 300

CUCM 8 is the first version supported in VMware. UCS hardware has to be used if you are using the VMware EXSi. VMware has to be EXSi 4.0 with SAN attached and it must use .ova template files in order to bring up the servers. Migration from hardware based to virtualized can be performed using disaster recovery DRF. You can use a mixed hardware cluster during migration. The License key for CUCM uses the VMware virtual MAC address.

For lab, there is a default 150 demo device license units included. with any CUCM startup desk. You can install up to 3 servers.

VMware Specifics

2virtual CPUs. 6GB RAM. Two 80 GB hard disks at SCSI id 0:0 and 0:1. The UCS version 7 VM is supported. Up to 7500 users per VM. No USB support. Backups must be via SFTP.

CUCM Server Roles:

CUCM Data base Publisher DB Server (Pub) can only be 1. Makes all the changes and sync to the sub servers. Only it can do some special services e.g. presence, Dir. Sync with the LDAP, MVA. Publisher can be used as a CPE call processing engine in a lab environment. Can have up to 8 subscriber DB servers running as CPE. Up to 3 CPEs per USM Group.

CUCM DB Subscriber server. Can have up to 2 subs running as tftp servers. Additional sub servers to be used as media servers e.g. MOH, Conference, Annunciator – plays a message when you do something, MTP. MTP and Conference can only speak g711 protocol. Avoid these even in a lab environment.

CUCM DB Roles:

IBM Informix DB is a structured query language DB (SQL). It is owned and managed by the Publisher server. There is a one way replication e.g. from pub to subs. Most config changes are made on the Pub DB server but there are some DB fields that can be changed on sub servers local to where the phone is registered and then replicated to the pub. During a Pub outage, the only features that can be changed are the UFF features. These are called User Facing Features e.g. Call Forward all, MWI, Privacy for shared lines enabled/ disabled, hunt group login and logout, device mobility, credential authentication and certificate authority proxy function status for end users and application users.

CUCM Cluster Communication:

ICCS – Inter cluster Communication Signaling. It uses TCP 8002-8004. IT replicates all run time data between all CPEs in a cluster. CPEs are the CUCMs that are running the call manager service.

CDR call detail records and CMR call management records. They are logged by the CPE where the phone or gateway is registered. This data is periodically pushed to publisher server. Cisco CAR CDR analysis and Reporting lacks in the amount of features. CAR and 3rd party billing apps servers always point to and collect from the pub. When using a 3rd party, we need to push the data to another server then query it to the pub.

CUCM Traffic flow:

Signaling between CUCM and phones or gateways – SIP/SCCP

Media does not flow through the CUCM Cluster. Endpoints e.g. Phones and gateways talk directly via RTP Media (voice / video).

Between the gateway and the PSTN – TDM Signaling/ media.


CCNA Voice Summary – Part 7

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Cisco CME (Cisco Unified Communication Express/ Call Manager Express)

Designed for enterprise branch offices and small medium businesses. It runs on a router. Supports a max of 450 IP phones but realistically 100 phones.  You can get voicemail using Unity Express (CUE) AIM card and the Network Module NM card. AIM – flash memory and smaller capacity.  Runs on Cisco ISR 2800, 2900, 3800 etc. Router can be configured via command line, Web UI and Cisco Configuration Professional Configuration CCP. It can be used to provide high availability fallback for large and branch offices using CUCM. This is Survivable Remote Site Telephony SRST. In SRST the phones fall back to operating on the local router. ME can provide 3rd party computer telephony integration. WE can use it as a single site or multi-site deployment. There is no clustering (active/active) but we can have HSRP as standby CME. CME is completely independent even during SRST. CME can run integrated directory and presence services. Interworks with CUCM. Does not integrate with CUPS. CME is just like a voice gateway with the ability to register phones on it. It can support remote phones

 CME Features:

It is a CPE. IT has divide control for VOIP endpoints with SIP and Skinny. Provides directory services, MOH server and conferencing server. Custom applications using TCL script e.g. BACD – basic auto attendant.

CME Traffic Flow.

SIP / SCCP signaling between the endpoints and CME only and not between endpoints directly.

RTP Media flows between the endpoints only.

CME Router Specs:

ISR- G2 – 2901 – 35 PHONES – 0 SM Service Module slots – up to 8 T1/E1

ISR- G2 – 2911 – 50 PHONES – 1 SM slots – up to 12 T1/E1

ISR- G2 – 2921 – 100 PHONES – 1 SM slots – up to 12 T1/E1

ISR- G2 – 2951 – 150 PHONES – 2 SM slots – up to 16 T1/E1

ISR- G2 – 3925 – 250 PHONES – 2 SM slots – up to 16 T1/E1

ISR- G2 – 3945 – 350 PHONES – 4 SM slots – up to 24 T1/E1

ISR- 1861 – 15 PHONES – 0 NM slots

ISR- 2801 – 25 PHONES – 0 NM slots

ISR- 2811 – 35 PHONES – 1 NM slots

ISR- 2821 – 50 PHONES – 1 NM slots

ISR- 2851 – 100 PHONES – 2 NM slots

ISR- 3825 – 175 PHONES – 2 NM slots

ISR- 3845 – 250 PHONES – 4 NM slots

RISR- 3250 – 20 PHONES – 0 NM slots

RISR- 3270 – 48 PHONES – 0 NM slots

Cisco Unified Communications 500 (UC500)

All in one unit. Built in PoE – up to 8 ports. Built in Router / Switch / Wireless / VPN / Firewall.

Unified Communication Manager Business Edition

Scalability up to 500 IP Phones. Combines Call Manager, Unity Connection and Unified Mobility into one box.

It has no redundancy.

CCNA Voice Summary – Part 6

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 Cisco UC Components:

Unified Call Control –CUCME or CME or UCME, CUCM or UCM or CCM

Unified Messaging – Unity Connection OR CUC or CUCx and Unity Express or CUE

Unified Presence – CUCM and CUPS (Cisco Unified Presence Server)

Unity is window based platform while Unity connection is Linux based.

Introduction to CUCM and CME

They are used for call processing.

Cisco Voice Infrastructure Model – multilayer and independent of each other. Consists of Endpoints; Applications like voice mail; call processing; Infrastructure.

The Core Products of CCNA Voice: CME; CUCM; Cisco Unity Connection – voicemail; Cisco Unity Presence – user tracking. you can track the status of users.

CCNA Voice Summary – Part 5

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Voice over IP

VOIP process: Sample analog Voice or Video ->; Encode to digital value ->; transmission in an IP data payload

Uses Signaling Protocol for setup and control of the call and a Media Protocol for the payload. They are transmitted separately.

Signaling is typically sent using TCP (connection oriented) because it is important that the other side gets all the signaling information.

After successful negotiation using a signaling protocol, the Media protocol takes over. Media protocol samples the voice/video using a codec and is sent over UDP (connectionless). Why UDP? – If voice does not get to the other side, is not very critical since we are sampling at a very high rate. Packet loss is not as critical as Packet delay.

The Voice gateway provides translation between different VoIP networks or VoIP and non-VoIP networks such as the PSTN. They also provide physical access for local analog and digital voice devices

VoIP trunks

  • easy to configure
  • Cost per line is less
  • Lower long distance rates
  • Data can utilize voice bandwidth when there is no voice
  • Usually SIP based

Cisco Unified Border Element (Cisco Multiservice IP-to-IP Gateway)

  • Demarcation point between two VoIP networks
  • Terminates and reoriginates both signaling and RTP/RTCP streams
  • Can interconnect VoIP networks using different signaling protocols
  • Provides NAT, Billing, CDR, QoS and media interworking.

Call Control Models:

Distributed – Each router is independent. Reduced scalability.

Centralized – Server / Client Model. Call Manager has all the intelligence. Simple model. Drawback – redundancy.

Design Models:

Campus (Single Site) – All Call Managers in one location. Usually uses the same single high bandwidth uncompressed codec. All application services are centrally connected. Connected to the PSTN and WAN. WAN is usually used for backup or connection to ITSP.

Central Site – all call processing agents in one single site. Can have connections to the remote office and the telecommuter via the WAN. You can also have a backup connection over the PSTN. Advantage TEHO – toll bypass. Remote routers can use SRST in case the central site is unavailable.

Distributed, Multi-cluster – sites are independent and with call processing. More expensive since we have to supply a full call processing model for each site.

CODEC – Compressor /Decompressor

A codec is a software algorithm that compresses and decompresses speech or audio signals.

Compressor IS used in the transmit direction / Decompressor is used in the Rx direction.

Codec is carried in the Media stream (RTP).


G.711 – ITU standard. Uses Pulse Code Modulation PCM (64KBS). Excellent audio. Typically used on LANS

G.722 – ITU standard. 64kbs. has been optimized for wideband speech. Sampling a wider spectrum of audio speech hence better audio quality than G.711

G.729 – Used for WAN connections. 8kbps. high complexity for CPU /DSP (for sampling audio). High complexity requires more chips/DSP channels to compress or decompress voice

g.729a – medium complexity cup/DSP

g.729b – high complexity codec. Added VAD (voice activity connection).added CNG (comfort voice generation). Most people disable g729b

iLBC – internet low bit rate codec. optimized for narrow band speech. 13.3kbps. originally designed by Skype and intended for the internet (lossy wan connections)

ISAC – internet speech audio codec. variable rate codec (ranges from 10 -32 kbps). it’s adaptive. optimized for wideband speech and jitter over WAN e.g. internet

Digital Signal Processors (DSPs).

DSPs Convert analog to digital and vice versa. They:

sample the analog voice ->; quantization of the sampled data ->;encode and packetize ->; optionally compress.

Packet Data Voice Modules (PVDMs)

This is cisco proprietary DSPs. Used in IP phones, VOIP gateways for voice termination, DSP Farms (DSPs grouped together) for conferencing and media termination point MTP or for Transcoding.

MTP terminates the media.

conferencing – mix together various audio/ video streams into one.

transcoder – change one codec into another. can also change the size of packets.

Streaming Protocols:

E.g RTP and RTCP.

Real Time Protocol.

Layer 4 protocol. encapsulates all delay sensitive traffic such as voice. Rides on top of UDP. UDP Ports 16384 – 32767.

Provides end-to-end network transport functions for delay senitive traffic

  • payload type identification
  • sequence numbering
  • time stamping
  • delivery monitoring

Real Time Control Protocol. Monitors quality and statistics for the RTP protocol. Uses a +1 port from the port used by RTP. RTP uses and odd port, RTCP uses even ports.

  • RTCP provides
  • session monitoring
  • session control
  • packet count, loss, delay, jitter

sRTCP – used for security. voice data payload is encrypted using the AES cipher using transport layer security TLS aka SSL v3.1. it does not encrypt the entire packet as IPsec does.

Encapsulation process:

Voice Payload ->; RTP Header ->; UDP header ->; Layer 3 IP header ->; Layer 2 .

Signaling Protocols:

Used to setup, teardown and control info about the call. Used for supplementary services like Call forward, pickup, transfer, hold, busy, redirect, call park, presence, Message waiting indicator MWI etc.

Most common: h323 – ITU ,sip – IETF ,MGCP – IETF, SCCP.


Evolved from ISDN Q931 layer 3 signaling .

Peer to peer protocols (phones and gateways have independent dial plans and are intelligent. they do not have to have a server in order to tell them what to do. they do not have to register to anyone but they have the ability to register to a gatekeeper so as to centralize dial plans.

SIP – Session Initiation protocol

Also a peer to peer protocol. Endpoints are intelligent and can make call routing decisions. They can register to a SIP Registrar server. it derives a lot of its roots from email SMTP.


MGCP- media gateway control protocol. designed for IP network to PSTN network voice gateways. from a TDM signaling protocol e.g. PRI to the IP network. it is a client/server (master /slave)protocol. gateway cannot at independently. client endpoints are not intelligent. clients/ slaves must register with the server.

SCCP – Skinny call control protocol.

Designed by Selsius systems who built call manager. SCCP was based on h323. h323 was too fat and had too many messages thus the skinner protocol.

Cisco proprietary. used for cisco devices e.g. IP phones, analog gateways, voice ports(unity and unity connection ) and gateways. skinny uses h225. it has q921 in its header messages.


CCNA Voice Summary – Part 4

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Circuit Switched Network

Involves 2 nodes that establish a dedicated circuit in order to communicate. The channel remains up during the whole conversation. PSTN is the largest Circuit switched network in the world. The PSTN uses SS7 signaling.

Advantages: Dedicated Channel; Excellent quality; delay and bit rate is constant

Disadvantages: Not always enough available channels; bitrate is limited – there is little chance for newer technologies to increase the sound quality; To enable Video, we need to bond channels together. e.g. h.320

 Packet Switched Network

Digital network that transmits data into packets irrespective of the data. Layer 3 devices encapsulate the data into layer 2 frames. Each device makes decisions for the packet independent of the previous device. Good Example is the Internet

Advantages: Constant bitrate; newer technology can improve sound, Easier to increase bandwidth for features. QoS overcomes all disadvantages

Disadvantages: No dedicated connection; Packet loss; Not always enough available bandwidth but can be added; Packet delay – latency; Jitter – delay variance.
Voice Packet Requirements

  • smooth bandwidth demand
  • minimal impact on other traffic
  • Small packets (60 – 200 bytes)
  • Delay intolerant
  • drop intollerant
  • UDP priority

Data Packet Requirements

  • varied bandwidth demand
  • can tolerate some delay
  • can tolerate high drop rates
  • use TCP for retransmission

Video Packet Requirements

  • bursty and greedy for bandwidth
  • impacts other traffic
  • Delay intolerant
  • drop intollerant
  • UDP priority

Unified Communication Networks

Everything flows over the same network as standard data. QoS keeps UC packets prioritized. They Include: Voice Calls, Conferences. VoiceMail, Presence and IM, Directory Services and phone based applications

 Public Providers

UCSystems can connect to both the old PSTN network and the Newer VoIP Network (SIP, H323, and ITSP – Internet Service Provider).

CCNA Voice Summary – Part 3

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Digital Connectivity

Problems with Analog connectivity:

  • Distance limitation – regeneration of the signal also leads to regeneration of the white noise
  • Wiring requirements – Analog requires a tip and ring wire for each call – that makes it 2 wires for each call.

These problems are overcome by digital signals. Digital signals converts the signals into zeros and ones. You can therefore go over any distance and can send the digital strings over one wire

Digitizing Voice

This process follows the Nyguist Theory which states – if you can sample over twice the highest frequency, you can accurately reconstruct a signal digitally. Nyguist theory samples between 300 – 4000 Hz. Human speech ranges between 200 – 9000 Hz. The Human ear can hear between 20 – 20000 Hz.

After sampling the signal. We perform Quantization of the sample. It involves taking the value of the amplitude of the analog signal and line it up to a specific value. This process is known as the Pulse Amplitude Modulation. Using PAM we take more samples at the lower levels since most of the human voice is close to the zero scale.

After Quantization comes Coding – change the value to binary numbers. This is referred to as Pulse Code Modulation PCM. There are 2 PCM methods (a-law and mu- law). Mu-law is used in the USA, Japan and Canada.

A-law is used in the rest of the world. It uses the first code value to represent positive or negative value. 0 – negative; 1- positive. Next 3 binary digits represent the segments. More samples are taken at the lower segments.. The next 4 values is the interval between the values in a given segment.

Mu-law; 0 – positive; 1 – negative. It Is the direct opposite of a-law.

To communicate between the two laws, we need to convert between the two standards.

After coding, we can optionally compress the samples. 8000 samples. Each sample is converted into 8 bits = 64000 bits per second per voice call. Compression saves on bandwidth.

During compression, you can choose to send just the changes between the sample or build a code book for each sample. G.729 consumes 8 kbps

Digital Voice Circuits

Carries information in channels: Bearer and Data Channels. Bearer – carries the voice sample, Data Channel – call control and signaling.

Types of Digital Circuits: Channel Associated Signaling (CAS) and Common Channel Signaling (CCS). CAS – voice and data are transported in the same channel (Robbed Bit signaling). CAS example T1/E1. CCS – the only thing in the bearer channel is the actual voice. there is a dedicated channel for signaling. CCS example T1 OR E1 ISDN PRI/BRI.

TDM is used in all circuits. Voice is sampled then divided and interwoven together.

BRI – 2bearer channels + 1data


CCNA Voice Summary – Part 2

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Analog Connectivity

Analog transmission – using some property of the transmission media to convey a signal. As you speak into an analog phone, the voice is converted into electricity. Analog transmission uses the properties of electricity to transmit signals.

 Analog signaling:

Loop start.

The phone has 2 connections – tip and ring. When the phone is off hook, the loop is connected and the signal passes through the circuit. Suitable for home use. When we have higher call volumes, this signaling would lead to glare. Glare – when you pick up the phone and you are bridged to the wrong incoming call.

Ground start

When the receiver is off hook, the phone shoots a ground signal grounds the ring wire temporarily hence receives a dial tone. Usually used for businesses.


Signaling types – supervisor signaling, informational and address signaling.

Supervisor signaling: used to send on hook, off hook and ringing. Ringing is sent using AC current instead of DC. DC current usually has to be looped back to the sending end. The AC does not require it to send back a signal.

Informational signaling – conveys information while you are on the phone. It is used to send dial tone, busy, ring back, congestion, reorder, receiver off hook, no such number and confirmation. These are electric signals that are being sent.

Address signaling – signals dialed digits. Types of Analog signaling – Pulse and DTMF Dual Tone Multi frequency.

Pulse –

DTMF – Each digit on the phone is assigned 2 frequencies to distinguish between the two.

Analog Voice Circuits

FXS – Foreign Exchange Station: Uses 2 wires (tip and ring) in an RJ-11 Port. Connects to the analog phones, Fax and FXO port. Provides power for the analog device and -48 V for ringing, voice progress tones and dial tone.

FXO – Foreign Exchange Office: Uses 2 wires (tip and ring) in an RJ-11 Port. Connects to the CO. Provides supervised disconnect and supports caller Id.

E&M (Earth and Magnet / Ear and Mouth): Uses 4 wires (the middle 2 are tip and ring, the outer ones are E&M that use out of band signaling). Connects to Tie line to PBX, Paging and MoH source.

CCNA Voice Summary – Part 1

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Introduction to VoIP

Why Use VOIP

Cost Savings: reduced wiring – one single cable for both phone and computer; reduced telecommuter costs since they connect to the HQ via internet; Free long distance between branches and HQ; Single inbox for voicemail, fax and email using unity connection; Saves office space using extension mobility; Open architecture – the use of standard compliant protocols.

Why Use WAN connection over PSTN Connection: Bandwidth compression over WAN; Free long distance; Get rid of the tie lies- get rid of the monthly cost.

How to Move to VOIP

New companies start VoIP systems from scratch. Older companies go in 2 phases: Phase 1 – Keep the old internal network but re-equip the voice gateways that will connect to the WAN and the PSTN; Phase 2 – Get rid of all the PBX systems and use end to end VoIP.

Traditional PBX (Post Branch Exchange)

This was a way of privatizing a company from the Central Office. It emulates the Central Office. It is divided into 2 sides: The Line/ Station side connects to the phones; The Trunk side connects to the Local PSTN, Central office or another PBX through tie-trunks. Everything in a PBX flows through a Time Division Multiplexing- TDM backplane. This causes a blocked architecture in PBXs whereby we cannot have all the phones in a large PBX site talking at the same time since there isn’t enough bandwidth.

Disadvantages of PBX Architecture

  • Dedicated separate lines for voice in PBX.
  • TDM Backplane caused congestion at peaks hours
  • Not enough channels for all phones attached to make a call.