Basic Voice over IP v1.01 – Aaron Balchunas
* * *
All original material copyright © 2007 by Aaron Balchunas (aaron@routeralley.com),
unless otherwise noted. All other material copyright © of their respective owners.
This material may be copied and used freely, but may not be altered or sold without the expressed written
consent of the owner of the above copyright. Updated material may be found at http://www.routeralley.com.
1
- Basic Voice over IP -
Voice over IP (VoIP)
Voice over IP (VoIP) is a digital form of transport for voice transmissions,
replacing analog phone systems.
The benefits of VoIP are considerable:
• Better use of bandwidth - Traditional voice requires a dedicated 64-
Kbps circuit for each voice call, while VoIP calls can use considerably
less. Additionally, no bandwidth is consumed when no call is being
made.
• Single form of cabling – Reduces implementation and maintenance
costs by having a standardized and consolidated cabling and
equipment infrastructure.
• Cost savings from integration into the data network – Toll charges
for inter-office voice communication can be avoided by routing voice
traffic across existing data lines.
• Integration into devices beyond telephones
Basic VoIP components can include:
• Phones – including both analog and IP phones.
• Gateways – allows a non-VoIP (analog) device to communicate with
the VoIP network, or a VoIP device to communicate with an analog
network.
• Application Servers – provides required applications to VoIP
phones.
• Gatekeepers – maps phone numbers to IP addresses, and grants
permission for call setup
• Call Agents – handles call routing and setup.
Digital Signal Processors (DSP’s) are used by devices to perform analogto-
digital and digital-to-analog conversions. Both VoIP phones and gateways
utilize DSP technology.
Basic Voice over IP v1.01 – Aaron Balchunas
* * *
All original material copyright © 2007 by Aaron Balchunas (aaron@routeralley.com),
unless otherwise noted. All other material copyright © of their respective owners.
This material may be copied and used freely, but may not be altered or sold without the expressed written
consent of the owner of the above copyright. Updated material may be found at http://www.routeralley.com.
2
VoIP Packetization
Voice traffic must be packetized as it traverses the IP network. Sound is
first captured using a microphone on the headset. A voice call requires a 4
kHz (4000 Hz) channel. To convert analog voice to a digital format,
samples of the frequency and amplitude of the analog wave are made. Thus,
sampling merely takes a snapshot of the signal at a given point in time.
The amplitude height of each snapshot is assigned a numeric value, through
a process called quantization. This numeric value is then represented as a
sequence of binary digits (usually 8) through a process called encoding.
The Nyquist sampling theorem dictates that the analog wave should be
sampled at a rate of twice the channel’s frequency range:
fs = 2(freq. range)
Thus, assuming a range of 4000 Hz, this requires a rate of 8000 samples per
second. Remember that each sample is assigned an 8-bit value to represent
the amplitude height at the time of sampling. Thus, a dedicated 64,000-bit
channel (8-bits x 8000 samples per second) was traditionally required for a
voice call (hence a DS0 being 64Kbps).
The process of encoding an analog signal into digital format is handled by a
codec (coder-decoder). The codec usually provides a level of compression.
The efficiency of the compression varies with the codec used; however,
more compression generally degrades sound quality. Various codecs
include:
• G.711 – uses 64 Kbps for a voice call
• G.726 – uses 32, 24, or 16 Kbps for a voice call
• G.728 – uses 16 Kbps for a voice call
• G.729 – uses 8 Kbps for a voice call
Generally, the analog sound is chopped into groups of 10ms, and then
sampled and encoded. Each group (or often two groups, for a total of 20ms
of analog sound) is encapsulated within an IP packet. At the transport layer,
Real-Time Protocol (RTP) is used instead of TCP. RTP operates on top of
UDP.
When the voice packet arrives at a digital-to-analog gateway, the headers are
stripped off, and the sound is reassembled as an analog stream.
Basic Voice over IP v1.01 – Aaron Balchunas
* * *
All original material copyright © 2007 by Aaron Balchunas (aaron@routeralley.com),
unless otherwise noted. All other material copyright © of their respective owners.
This material may be copied and used freely, but may not be altered or sold without the expressed written
consent of the owner of the above copyright. Updated material may be found at http://www.routeralley.com.
3
Cisco VoIP Integration
Cisco devices operating as VoIP gateways can contain a variety of analog
interfaces, including:
• Foreign Exchange Station (FXS) interface – connects to an analog
device, providing the appropriate voltage and dial tone.
• Foreign Exchange Office (FXO) interface – connects to a PBX
(Private Branch Exchange) or PSTN (Public Switched Telephone
network).
• E&M interface – can also be used to connect to a PBX, or is used for
PBX-to-PBX connections.
Additionally, Cisco gateways can connect to provider PBX’s and networks
using digital interfaces, including:
• ISDN BRI and PRI
• T1/E1 CCS (Common Channel Signaling) – employs a dedicated
channel for signaling.
• T1/E1 CAS (Channel Associated Signaling) – a portion of each
channel is utilized for signaling.
Basic Voice over IP v1.01 – Aaron Balchunas
* * *
All original material copyright © 2007 by Aaron Balchunas (aaron@routeralley.com),
unless otherwise noted. All other material copyright © of their respective owners.
This material may be copied and used freely, but may not be altered or sold without the expressed written
consent of the owner of the above copyright. Updated material may be found at http://www.routeralley.com.
4
VoIP Signaling Protocols
VoIP protocols are responsible for the three key stages of a voice call:
• Call setup
• Call maintenance
• Call teardown
The most common VoIP protocols are as follows:
• H.323 – an ITU standard
• Session Initiation Protocol (SIP) – an IETF standard
• Media Gateway Control Protocol (MGCP) – an IETF standard
• Skinny Client Control Protocol (SCCP) – Cisco proprietary
Basic Voice over IP v1.01 – Aaron Balchunas
* * *
All original material copyright © 2007 by Aaron Balchunas (aaron@routeralley.com),
unless otherwise noted. All other material copyright © of their respective owners.
This material may be copied and used freely, but may not be altered or sold without the expressed written
consent of the owner of the above copyright. Updated material may be found at http://www.routeralley.com.
5
Cisco VoIP CallManager Configuration – Voice Trunk Ports
Traditionally, Cisco IP phones contain a switch with two interfaces. The first
interface connects the IP phone to the wall jack. The second interface
connects the user’s workstation to the IP phone. This allows a single cable to
handle the user’s voice and data needs.
To keep the voice/data traffic segregated, the IP phone forms a trunk link
with the remote switch. Data traffic is tagged as a different VLAN than the
voice traffic. Configuration on the remote switch (or the call-manager
functioning as the switch) is simple:
VoIP-Switch(config)# interface FastEthernet0/1/4
VoIP-Switch(config-if)# switchport access vlan 50
VoIP-Switch(config-if)# switchport trunk native vlan 50
VoIP-Switch(config-if)# switchport mode trunk
VoIP-Switch(config-if)# switchport voice vlan 60
In the above example, data traffic will be tagged as VLAN 50, while voice
traffic will be tagged as VLAN 60.
Basic Voice over IP v1.01 – Aaron Balchunas
* * *
All original material copyright © 2007 by Aaron Balchunas (aaron@routeralley.com),
unless otherwise noted. All other material copyright © of their respective owners.
This material may be copied and used freely, but may not be altered or sold without the expressed written
consent of the owner of the above copyright. Updated material may be found at http://www.routeralley.com.
6
Cisco VoIP CallManager Configuration – Dial Peers
Dial Peers provide call-routing, and serve two key functions:
• VoIP dial-peers - used to connect Cisco call-managers, gateways,
or gatekeepers to other such VoIP devices. For example, two callmanagers
at separate branches would point to each other using
VoIP dial-peer commands.
• POTS dial-peers – used to connect Cisco VoIP devices to an
analog device or network. A dial-string is mapped to a local analog
port on the VoIP gateway or call-manager.
Thus, the function of a Dial Peer is to match an incoming call with a
destination pattern, which points to either a remote device or local interface.
To configure a VoIP dial-peer:
CallManager(config)# dial-peer voice 1 voip
CallManager(config-dial-peer)# session procotol sipv2
CallManager(config-dial-peer)# session target ipv4:10.1.5.50
CallManager(config-dial-peer)# destination-pattern 15865551212
CallManager(config-dial-peer)# codec g711ulaw
The above configuration maps a sip connection to a remote VoIP peer at
address 10.1.5.50 for phone number 15865551212. The g711 codec is being
employed.
To configure a POTS dial-peer:
CallManager(config)# dial-peer voice 2 pots
CallManager(config-dial-peer)# destination-pattern 1212
CallManager(config-dial-peer)# port 0/2/0
The above configuration maps an extension or phone number of 1212 to the
analog voice port 0/2/0.
(Reference: http://www.cisco.com/univercd/cc/td/doc/product/software/ios123/123cgcr/vvfax_c/int_c/dpeer_c/dp_ovrvw.htm)
Basic Voice over IP v1.01 – Aaron Balchunas
* * *
All original material copyright © 2007 by Aaron Balchunas (aaron@routeralley.com),
unless otherwise noted. All other material copyright © of their respective owners.
This material may be copied and used freely, but may not be altered or sold without the expressed written
consent of the owner of the above copyright. Updated material may be found at http://www.routeralley.com.
7
Cisco VoIP CallManager Configuration – Telephony Service
Telephony-Service configuration provides a wide variety of global
configuration options for a Cisco CallManger:
CallManager(config)# telephony-service
CallManager(config-telephony-service)#
The configuration files for specific models of IP phones are stored in flash,
with a .bin extension. To load these configuration files:
CallManager(config-telephony-service)# load 7914 S00105000200
CallManager(config-telephony-service)# load 7920 7920.4.0-02-00
CallManager(config-telephony-service)# load 7960-7940 P00308000400
To specify the maximum number of phones that can register with the Call-
Manager (dependent on the hardware/software platform):
CallManager(config-telephony-service)# max-ephones 12
To specify the maximum number of directory numbers (DNs) the Call-
Manager will support (also dependent on the hardware/software platform):
CallManager(config-telephony-service)# max-dn 48
To specify the IP address of the Call-Manager on the voice VLAN:
CallManager(config-telephony-service)# ip source-address 10.5.5.1 port 2000
To specify the extension for voicemail:
CallManager(config-telephony-service)# voicemail 2000
To specify the audio file for music-on-hold:
CallManager(config-telephony-service)# moh music-on-hold.au
To configure a username and password for the Call Manager’s web
interface, and to enable configuration of DN’s through that interface:
CallManager(config-telephony-service)# web admin system name AARON password CISCO
CallManager(config-telephony-service)# dn-webedit
To access the web-interface, use the following URL:
http://IPADDRESS/telephony_service.html
Basic Voice over IP v1.01 – Aaron Balchunas
* * *
All original material copyright © 2007 by Aaron Balchunas (aaron@routeralley.com),
unless otherwise noted. All other material copyright © of their respective owners.
This material may be copied and used freely, but may not be altered or sold without the expressed written
consent of the owner of the above copyright. Updated material may be found at http://www.routeralley.com.
8
Cisco VoIP CallManager Configuration – Telephony Service (continued)
To allow the transferring of calls to an outside line, using a specific dialpattern
(such as dialing 9 first, and then a seven-digit number):
CallManager(config-telephony-service)# transfer-pattern 9 . . . . . . .
To configure the auto-attendant for night-service:
CallManager(config-telephony-service)# night-service code *11
CallManager(config-telephony-service)# night-service day Mon 18:00 06:00
CallManager(config-telephony-service)# night-service day Tue 18:00 06:00
CallManager(config-telephony-service)# night-service day Wed 18:00 06:00
CallManager(config-telephony-service)# night-service day Thu 18:00 06:00
CallManager(config-telephony-service)# night-service day Fri 18:00 06:00
To configure a directory of extensions:
CallManager(config-telephony-service)# directory entry 1 3000 name Aaron
CallManager(config-telephony-service)# directory entry 1 3001 name Petey
CallManager(config-telephony-service)# directory entry 1 3002 name Team Awesome
CallManager(config-telephony-service)# directory entry 1 3003 name Team Tiger
CallManager(config-telephony-service)# directory entry 1 3003 name Jack Nicholson
CallManager(config-telephony-service)# directory entry 1 3004 name Nick Cage
To define the URL’s for ephones:
CallManager(config-telephony-service)# url directories http://10.5.5.1/localdirectory
CallManager(config-telephony-service)# url services http://10.5.5.1/menu.php
CallManager(config-telephony-service)# url authentication http://10.5.5.1/auth.php
Some configuration changes require a reset of the phone(s). To reset all
phones connected to the Call Manager:
CallManager(config-telephony-service)# reset all
(Reference: http://www.cisco.com/en/US/tech/tk1077/technologies_configuration_example09186a00800ffdcc.shtml)
Basic Voice over IP v1.01 – Aaron Balchunas
* * *
All original material copyright © 2007 by Aaron Balchunas (aaron@routeralley.com),
unless otherwise noted. All other material copyright © of their respective owners.
This material may be copied and used freely, but may not be altered or sold without the expressed written
consent of the owner of the above copyright. Updated material may be found at http://www.routeralley.com.
9
Cisco VoIP CallManager Configuration – DN’s and Extensions
Directory Numbers (DNs) are assigned to phones for identification and to
allow call-routing. Call Managers usually support a finite number of DNs,
depending on the hardware/software/licensing platform.
Extensions are then mapped to these DNs. The Call Manager identifies the
phone using a DN, but users call a phone using the extension number. To
configure a DN:
CallManager(config)# ephone-dn 5 dual-line
CallManager(config-ephone-dn)# number 3001
CallManager(config-ephone-dn)# description Call at your own risk
CallManager(config-ephone-dn)# name Petey
In the above example, dn 5 is configured as a dual-line, which allows for
call transfer, conferencing, and call waiting. DNs/extensions that serve
solely as a voicemail box can leave off the dual-line parameter.
An extension number of 3001 has been assigned to this DN.
(Reference: http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122z/122zj15/cme30cr/icr30am.htm#wp1008071)
Cisco VoIP CallManager Configuration – Phones
To configure the actual VoIP phone:
CallManager(config)# ephone 7
CallManager(config-ephone)# mac-address 0011.2233.4455
CallManager(config-ephone)# type 7920
CallManager(config-ephone)# button 1:5
CallManager(config-ephone)# pin 12345
CallManager(config-ephone)# speed-dial 1 3002 label “Team Awesome”
In the above example, ephone 7 has been identified as having a mac-address
of 0011.2233.4455, and a model number of 7920.
The button command maps the first button (or extension on the phone –
Cisco phones support multiple extensions) with DN number 5 (per our
previous configuration, this maps to extension 3001).
Finally, the phone’s pin number for voice mail has been set to 12345, and a
speed-dial entry has been added for extension 3002.
(Feference: http://www.cisco.com/univercd/cc/td/doc/product/software/ios122/122newft/122limit/122z/122zj15/cme30cr/icr30am.htm#wp1014674)