Voip Technologies: Study &
Implementation
, Laboratoire SUPINFO des Technologies Cisco
http://www.supinfo.com
Voip Technologies: Study & Implementation
by
Voip Technologies: Study & Implementation
This
program
will
speak
about
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VoIP
technologies.
Chapter 1. Introduction to VoIP
Course Objectives:
Discover the VoIP
Know the advantages and Drawbacks of VoIP
Know the different kind of use of VoIP
1. Course
1.1. VoIP, ToIP and the traditional Telephony
Sometimes called VoIP (Voice over IP) or ToIP (Telephony over IP), the voice transmission over IT networks is
the result of combined permanent needs in communication of our society, and democratization of these data
networks. These ones offer an improved reliability for data transport and Internet connection offers increasingly
more accessible and interesting.
There is a notable difference between VoIP and ToIP. VoIP means real-time flow transport, especially voice, on
data networks. VoIP becomes ToIP when it is linked to the traditional telephony network. Thus, ToIP is like a
subset of the VoIP, with some constraints imposed by the world of traditional telephony, in particular to respect
phone number’s nomenclature.
The objective of VoIP is replace, at least for a part, the traditional telephony often expensive, mainly for
international calls, by using data networks deployed around the world.
The IT networks being built for the data transmission; the voice is thus digitized, through the use of a codec, and
then encapsulated into a packet before being transported. The codec used for IP telephony don’t really depend
on the protocol used, but rather of the support on software and network devices.
The voice transportation on a digital form can be from end to end or more generally partial. In this last case, the
international telephone calls might be reduced to local communication cost. Corporation like Skype have widely
contributed to demonstrate this real interest on cost killing.
The voice digitalization exists for a very long time, and vocal exchanges through computers are not rare since
instant messaging for example.
There was a too large barrier between the IT world and the world of traditional telephony. The VoIP has really
appeared when we began to make the link between data-processing and traditional telephony, jointly to the
development and ratification of specific norms and protocols.
VoIP exists since many years within companies, depending on existing technology and financial means. A good
example is the Cisco Systems company, which has placed from the beginning their own products and services
for internal use. Besides, some providers sell their Voice on IP services for more than 10 years.
Just like any other new technologies, VoIP increases in popularity when the prices of products had been
significantly reduced. The ISP (Internet Service Provider) offer including telephony service has also greatly
contributed to cut out VoIP marginalization for personal use.
1.2. Advantages and Drawbacks
1.2.1. Advantages
The growing of IP telephony is obvious. This is mainly due to advantages that this technology provides
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compared to traditional telephony.
Among these advantages, we can see the followings:
Mobility
Unique architecture
Economies
Added services
Mobility: Unfortunately, traditional office telephones can only be present in one location at a time. IP
telephones can follow the user whatever he goes, the only real constraint being to have access to the data
network. In addition, it is possible to possess multiple IP terminals for the same user (one IP telephone on the
office and one Softphone with VPN access when traveling for example), a protocol working on background to
manage the current location of the user and the call redirection to the good terminal.
Unique architecture: One of the objectives of the IP telephony is to integrate the telephone network to the data
network, in order to form a single and unique network. Moreover, it is possible to use only one link to a service
provider to transport data (Internet connection), whereas before a traditional telephone link was mandatory.
Economies: Another advantage is the cost of the calls. Indeed, a well designed VoIP infrastructure provides
lower communication costs, even null in certain cases, whether they are local or international. The possibility of
not subscribing to a traditional telephone link permits to do substantial economies.
Added services: The most important advantage of VoIP, except the costs, is the virtually unlimited list of
services that can be added to it. We can quickly list these ones:
Voice Mail: Answering machine synchronized with mail servers, to receive voice messages on emails.
Click-to-Dial: Telephone call initiation through an email client (example: Microsoft Outlook).
Presence management: Automatic redirection to the nearest terminal from the user.
Contact synchronization: Centralization of postal addresses, emails and telephone numbers inside a single
directory.
1.2.2. Drawbacks
Some advantages may also appear as drawbacks, depending of the case. Thus, we can list the following
drawbacks:
Unique architecture
Cost of the VoIP
Quality and reliability
Unique architecture: Combining networks together can create problems that shouldn’t exist before. Indeed, the
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VoIP is an application deployed on a standard network infrastructure just like any other application. Some
details must then be taken in account (like QoS, security of transmissions, availability, and strength against
network strikes, etc.), in order to guarantee the telephone service.
Cost of the VoIP: Unfortunately, the cost of VoIP is mainly linked to the infrastructure and devices used.
That’s why some corporations wouldn’t gain something when migrating to VoIP. In general, corporations
choose for a progressive transition to VoIP in order to replace, in the long term, the traditional telephones and
PABX. This progressive transition is done by using adapters, mainly FXS and FXO.
Consequently, except some exceptions, only the brand new corporations or branches choose a complete IP
solution, because the data network can be correctly built to support VoIP from the beginning. The other
corporations would rather prefer a progressive transition.
Quality and reliability: The IP telephony using the data networks, Internet included, to transport the flows, the
calls may then undergo some disagreements (packet loss, delay, etc.) harming the general quality of the
communication.
For example, it is generally well known that a delay below 150 ms is required for an optimal quality. However,it
is unfortunately current on certain connections to exceed this delay (medium delay observed on a satellite link is
around 500 ms).
Moreover, the VoIP being an application traveling across the network, it is then as sensitive as any other
application to problems that can happen on this network, like denial of services (DoS and DDoS) or more
simply to congestion.
1.3. Kinds of use of VoIP
1.3.1. VoIP at Home
More and more, ISP provides a VoIP connection at home. Most of the time, the VoIP is invisible for the
customers, they don’t see any changes, they continue to use their analogical phone. But in fact, they use VoIP.
They just plug their phone into the router provide by Internet Service Provider (ISP) . With this way, the
customers can reduce his cost without any change. Some provider begins to supply a VoIP WiFi channel to
allow your mobile phone to use your house phone by WiFi. The goal of that is to increase the roaming of the
VoIP. You are at home; you don’t need to care about it, all your communication will be through your internet
connection, if you leave your house, you will use your GSM (or the 3G).
1.3.2. VoIP for professionals
As you know a company can’t work without communication, for most of them the phone is their principal tool.
To reduce cost and increase the potential of this one, they use more and more VoIP. They can provide a phone
(virtual or not) for all employees, and allow them to use it to communicate between them.
i.e. We have two branch in a company, one in Paris, the other one in Beijing. The employees of this two branch
can communicate between them. They reduce the cost of the company, and also increase the communication
efficient.
1.4. Providers, devices and services
The IP telephony market is quite wide. Numerous corporations have invested on this market, by proposing their
solutions.
The actual well known names in the VoIP world are:
Alcatel
Audiocode
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Cirpack
Cisco
Linksys
Quintum
RAD
Moreover, multiple providers offer their services linked to VoIP. To draw up a list these providers should be
impossible, taking into account their number. On the other hand, we can differentiate them in several categories:
ISP providing VoIP services: All or nearly almost, the classical Internet providers now offer VoIP services
to their clients. We can for example list Telecom Italia, or any other ISP that provides telephone service
through boxes.
Dedicated VoIP providers for corporations: These providers, sometimes quite new on the VoIP market, are
dedicated to offer VoIP solutions to corporations (like Verizon).
Dedicated VoIP providers for personal use: These providers (Skype, Vonage, VoIP Buster, etc.) generally
only provide access to the VoIP infrastructure, the flow transiting across a classical Internet connection.
These solutions are thus limited in functionality and quality of service. They are reserved for personal use.
1.5. The next step of the VoIP
VoIP will be able to profit of capacities and virtually unlimited features, as additional services to the simple call,
provided by IT networks. The only limit to these services is the imagination of software developers.
The future of VoIP is clearly oriented to centralization and multiplication of offered services (IP Centrex
platforms), but also to the increased mobility and interoperability with 3G/Wi-Fi systems (« dual mode »
smartphones).
The IP Centrex solutions begin to be widely available, through multiples platforms. IP Centrex globally
corresponds to a centralized service offer based on a single platform, hosted at the provider’s datacenter.
The protocols actually in place allow a great flexibility in their use. So, it is theorically possible to let any kind
of real-time flow travel on the VoIP network, the limit is on implementation of the corresponding features into
softwares and service platforms.
Moreover, the new smartphones, having a Wi-Fi interface, allow « dual mode » communications. This system
provides the best way to communicate whatever the available environment. If a Wi-Fi hot-spot is near, the VoIP
client can then be used. If not, the telephone can automatically switch to the GSM network.
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Chapter 2. Protocols
Course Objectives:
Know the different protocols used in VoIP
Know the different architectures
1. Course
1.1. Signaling and Flow Transport
Lots of protocols are, or were, used to transmit real-time flows over IP networks. It is possible to distinguish two
kind protocol families:
Signaling protocols
Used for call establishment and control.
These protocols can be then divided by their working process:
Client / Server (asymmetric)
Peer-to-Peer (symmetric)
Table 2.1. Comparative of protocols
Protocol
Client/Server
Cisco SCCP
X
MGCP
X
Peer-to-Peer
SIP
X
H.323
X
Flow transport protocols
Permit flow traval across the IP network
1.1.1. Signaling Protocols: Client / Server
1.1.1.1. MGCP protocol
MGCP (Media Gateway Control Protocol) is a signaling protocol specified in January 2003 by the RFC 3435,
and is based on the RFC 3015. It has been developed jointly by the MEGACO work group of the IETF and the
ITU-T (International Telecommunication Union – Telecom standardization).
MGCP is generally encapsulated in UDP segments, on port number 2427. It uses SDP to describe the media and
RTP to transport the flows.
Figure 2.1. MGCP architecture
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MGCP is a client/server architecture composed by the following elements:
Call Agent
It’s the softswitch of the VoIP network with MGCP. Its main purpose is to coordinate MG and SG, by
indicating them which one has to accomplish the required functions. It is the Call Agent that receives the
notifications and redirects the functions of the VoIP infrastructure.
Media Gateway (MG)
At least one MG has to be present in the MGCP architecture. It mainly focuses on flow conversion between
circuits (TDM) and packets (IP). More generally, it works on the flows.
Signaling Gateway (SG)
At least one SG has to be present in the MGCP architecture if there is a connection with another telephone
network. It permits to convert signaling information (calls, etc.) from and to another network.
Endpoint
It is the source of signal. This can be a telephone, a conference server or an interactive voice response (IVR)
server for example.
It can have multiple Call Agents in a single infrastructure, thus guaranteeing availability through redundancy
and load balancing of call management.
Then, the different functions of the VoIP network are delegated to one or more MGCP gateways (MG and/or
SG).
It’s possible to centralize multiple functions in a single network device. So, it’s not rare to see servers that
integrate a Call Agent, a MG, a SG, and an endpoint (like a conferencing server).
1.1.1.2. Cisco SCCP protocol
SCCP is a Cisco protocol employed for the communications in real time as well as the conferences. It’s used by
its CallManager platform and its IP phones.
The protocol is built around a server, that can be a cluster, and that centralize the entire processing, thus
simplifying functions managed by terminals (telephones and IP/TDM gateways).
The centralized functions come from simple call processing, to updates of terminal firmwares, through
providing multiple services.
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The advantages of protocol SCCP rest on its weak requirements in memory and load processor. The protocol
can be employed within the framework of a made safe LAN, with a quality of sufficient band-width.
One of the disadvantages of SCCP is the management of QoS as that of the bandwidth which aren’t taken into
account by the protocol. In the same way, protocol CRTP (Compressed Real-time Protocol Transport) is not
supported. SCCP has its limits and does not allow either to authenticate distant users, out of the LAN of CME.
Figure 2.2. SCCP protocol
In spite of the use of a connection VPN, protocol SCCP remains unable to manage the distant users. Each site
must have a router Cisco CME to authenticate phones IP locally. Operation through WAN between several
routers CME is carried out then by the means of the H.323 protocol.
1.1.2. Signaling Protocols: Peer-to-Peer
1.1.2.1. SIP protocol
SIP (Session Initiation Protocol) is a protocol at the application layer (layer 7) of the OSI model. It has been
drafted by the MMUSIC (Multiparty Multimedia Session Control) group of the IETF (Internet Engineering Task
Force) in March 1999. The guideline was to conceive an easy to implement, scalable and adaptable signaling
protocol. In June 2002, a new version of the norm, the RFC 3261, is published. It represents today the
fundamentals of the SIPv2 protocol.
The purpose of SIP is to establish, modify and terminate multimedia sessions with one or multiple participants,
independently to transport layer protocols and without dependencies on the type of established session. A
participant can also be invited into a pre-established session. In the same way, a data would be added or deleted
from an existing session.
A session is a subset of callers and callees that communicate together. Multimedia conferences and phone calls
through Internet are examples of sessions.
However, SIP is not the only protocol required by communication devices. Indeed, its goal is to make the
communication possible, the communication by itself has to be done via another way. This implies that SIP has
to be combined with other protocols to obtain a complete multimedia platform.
Typically, following the RFC 3261, this implies the following protocols:
RTP (Real-time Transport Protocol)
To ensure the transport of real-time flows. It encodes and divides data into packets, and then transports them
across the IP network.
SDP (Session Description Protocol)
To describe multimedia session parameters.
RTSP (Real-Time Streaming Protocol)
To control delivery of streamed flows.
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MGCP (Media Gateway Control Protocol)
For gateways used to control access to the public telephone network (POTS).
RTP and SDP are the most often used protocols with SIP.
SIP is based on the HTTP protocol, which is also considered as a signaling protocol because it permits to
request a specific resource to a server. SIP profits of the tested value of the protocol undoubtedly the most used
and recognized in the world.
SIP can be transported into TCP or UDP segments. The default port number is 5060, except when using TLS
(Transport Layer Security) which use de default port number 5061.
1.1.2.2. H.323 protocol
H.323 is a protocol suite worked out by the ITU-T, defining standards for the multimedia communications. The
first draft was done in 1996, and the actual version (version H.323v6) arrived in June 2006.
The described protocols are grouped in seven categories:
Call control and signaling
H.225.0
Call signaling protocols and packetization of multimedia flows (uses a subset of Q.931 signaling)
H.225.0 / RAS
Registration, Admission and Status
H.245
Control protocol for multimedia communication
Audio codecs
G.711
G.722
G.723.1
G.728
G.729
Video codecs
H.261
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H.263
Data transmission
T.120
Protocol suite for data transmission (used by real-time collaboration applications)
Transportation on the media
RTC
RTCP
Security
H.235
Security and encryption for multimedia terminals (H series)
Additional services
H.450.1: Generic functions for additional services
H.450.2: Call transfer
H.450.3: Call diversion
H.450.4: Call hold
H.450.5: Call park and pick-up
H.450.6: Call waiting
H.450.7: Message waiting indication
H.450.8: Name identification services
H.450.9: Call completion services for H.323 networks
The H.323 communications don’t necessary need a central point. We are in a Peer-to-Peer way of working. So,
an H.323 terminal can directly communicate with another H.323 terminal without going across a server.
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Figure 2.3. H.323 architecture
The H.323 architecture is composed by:
Terminals
Describes the endpoint for each link. It provides two methods for real-time communication with another
H.323 terminal, a gateway or a MCU. This communication is composed by a combination of dialogs, data
and/or videos.
Gateways
They establish the connection between H.323 terminals, as well as with terminals using other protocols like
POTS, SIP or MGCP.
Gatekeepers
Provide registering and authentication mechanisms to terminals, permits bandwidth control, ensure translation
between telephone number and IP address, and also call transfer for example.
MCUs (Multipoint Control Unit)
They establish conferences and are composed by:
Multipoint Control elected to ensure call signalization and conference control.
Multipoint Processor that provides communication and flow mixing. Sometimes, it ensures transcribing of
received audio and video flows.
1.1.3. Flow transport protocols
1.1.3.1. RTP protocol
RTP, for Real-Time Transport Protocol, has been developed in January 1996. The last version is written in the
RFC 3550 published in July 2003.
RTP provides end-to-end transport functions adapted to real-time data applications, like audio, video or data
simulation flows, across unicast or multicast delivery services. RTP works with RTCP, which is in charge of the
QoS and information transportation about participants in a current session.
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The RTP data is typically transported into UDP segments, application using RTP being generally very few
sensitive to packet loss but more on latency. There is no default port number, except for a detail: the RTP flow
uses a pair port number and the associated RTCP flow uses the odd port number that follows (RTCP port = RTP
port + 1).
The services proposed by RTP are the followings:
Identification of the load type
Sequance numbering
Time stamping
Delivery monitoring
The convenient delivery mechanisms and other QoS guarantees are not done by RTP, but by lower layer
protocols. Consequently, RTP believes that the network is reliable.
Numerous applications are built to use RTP. Here are some examples:
Multimedia conferencing with multiple participants
Continuous data storage
Distributed interactive simulation
Active badge for room access
Control and measurements applications
1.1.3.2. RTCP protocol
RTCP (Real-Time Transport Control Protocol) is an integral part of the RFC 3550 which also define RTP.
This protocol provides different periodic, out-of-band control services for RTP flows. It’s an out-of-band flow
because RTP and RTCP packets are distinct.
RTCP provides four main functions:
Feedback about transmission quality: This is done by using source reports (SR) and receiver reports (RR).
Transportation of the source RTP flow identifier (CNAME): Quite useful for example when a receiver
has to associate 2 different RTP flows for the same session (voice and video flows of a visioconference).
RTCP packets sent by all the participants: This feature permits to all the participants to observe the
number of participants in a multimedia session. This also permits to calculate the frequency of sending
reports (SR and/or RR) to adapt itself to any number of participants.
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Minimal control information for a session (optional): This information provides details about the arrival
and departure of participants in a conference. It’s then possible, for example, to maintain a real-time list of
participants in this conference.
Using RTCP is not mandatory but strongly recommended for any session, especially for the ones in a multicast
environment (multiple participants).
1.1.3.3. SRTP protocol
The SRTP protocol, for Secure RTP, is an evolution of the RFC 3550 answering to authentication
confidentiality and integrity problems of transported flows. It’s defined jointly to SRTCP (Secure RTCP) in the
RFC 3711 published in March 2004.
The confidentiality is ensured by the AES algorithm, which can be implemented via different cipher modes.
The authentication and integrity as well as replay protection are ensured by the HMAC-SHA1 algorithm (160
bits fingerprint).
The functions of RTP and RTCP are the same in the secured versions.
1.2. Codecs
A codec is a compression/decompression algorithm used to represent a signal, generally audio or video, in a
digital environment.
It exist multiple codecs, each one having differences mainly on the quality of the compressed signal, represented
by a MOS (Mean Opinion Score) index, and the calculation load to process the signals.
These codecs have multiple criteria in common:
Bit rate
Sample size (typically 20 or 30 ms)
Latency due to sampling (equal to the sample size)
Compression/decompression time for the signal (variable latency depending of the algorithm)
Number of frames per second
Here is a table grouping the most used codecs in the VoIP world, their bit rate, and their used bandwidth on an
Ethernet network (believing that 40 bytes are necessary for the sum of UDP, IP and Ethernet headers):
Table 2.2. Used bandwidth for different codecs
Codec
Codect Bit Rate
Bandwidth on an Ethernet
network
G.711
64 Kbps
87.2 Kbps
G.723.1
5.3 Kbps
20.8 Kbps
G.723.1
6.4 Kbps
21.9 Kbps
G.726
24 Kbps
47.2 Kbps
G.726
32 Kbps
55.2 Kbps
G.728
16 Kbps
31.5 Kbps
G.729
8 Kbps
31.2 Kbps
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So, we can estimate the maximum number of simultaneous calls that can be managed on a link.
A bandwidth calculator is available at this URL: http://www.bandcalc.com
The throughput from the previous table is for one flow. A telephone communication generally uses two flows,
the sending and the receiving, and then we have to take a great care on consequent link choice.
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Chapter 3. Equipment
Course Objectives:
Discover the several equipment use in VoIP
Know the differences between them
Know what is a Gateway and a Gatekeeper
1. Course
1.1. IP Phones
An IP phone is a telephone terminal that connects to a network device in place of on a standard telephone jack.
Thus, all the telephone communications will no longer travel on a standard telephone line but on a data network.
However, it exist two types of IP phones:
Wired IP phones
Wireless IP phones
Figure 3.1. Cisco IP Phone 7970G and Zyxel P2000W
1.2. IP Telephony Softwares
The IP telephony softwares, also called « Softphones », permit to call via a computer provided with a headset
and microphone, just like if it is a physical telephone with the same features.
It exist numerous IP telephony softwares, among the most popular we find: Skype, Live Messenger,
Counterpath eyeBeam, and many more.
The features and supported protocols depend on the software. So, we have to choose the software based on the
platform used.
Figure 3.2. CounterPath eyeBeam
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1.3. Analog Telephone Adapter (ATA)
In order to permit a progressive change to a VoIP infrastructure, FXS type adapters have arrived to connect
standard telephones on an IPBX, or conversely to plug a standard telephone architecture (telephones and PABX)
to an IP network.
Figure 3.3. Cisco ATA 186 and Linksys PAP2
The FXO ports are generally located on IP/TDM gateways.
The FXS (Foreign eXchange Subscriber) and FXO (Foreign eXchange Office) ports are the interfaces on a
standard telephone network. The FXO port is the downstream interface (going from the telephone to the PABX
for example), whereas the FXS port is the upstream interface (going from the PABX to the telephone).
Figure 3.4. Exhibit of FXO and FXS at the customer side
The FXS port provides to the subscriber tone and electric power supply services. It’s the port that goes to the
subscriber.
The FXO port mainly provides the local loop termination service by indicating if the headset is hung up or not
(on-hook/off-hook). It’s the port that goes to the provider (Telco).
1.4. PABX and IPBX
A PABX (Private Automatic Branch eXchange), or PBX (Private Branch eXchange), is a standard telephone
switch. This device interconnects multiple standard telephone terminals of a corporation. It provides multiple
services like call forward and standby music.
It also exists virtual PABX, also called IPBX (Intranet Private Branch eXchange), which are mainly the same
than PABX but for a VoIP use. These IPBX are used for example by IP Centrex platforms.
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1.5. Cisco Call Manager Express (CCME)
CCME (Cisco CallManger Express train) is a solution developed by Cisco for the routers. It allows them to
manage the calls and offer different customer services of the network. It is integrated into the router IOS.
This solution makes possible to set up an economic VoIP solution with high reliability and without the
complexity of deployment of a solution based on servers in a small structure. By this way, it offers the VoIP
services for approximately 300 users and 120 IP Phones.
Also this solution is compatible with most of management solutions for networks, and theirs materials like
routers, switches, gateways, etc...
1.6. Gateways
A gateway is a device. It allows converting classic telephony traffic into VoIP. Gateways are used in 2 ways: To
convert incoming telephony traffic to the VoIP line, and to internet-connect VoIP networks.
Optional features can be added to a gateway such as Gatekeepers, billing systems, Softswitches, and network
management systems.
A Gatekeeper is use to manage Gateways and MCU’s (multipoint control units) and also to make the routing.
The gatekeeper can implement security policies on gateways, improve the QoS (Quality of Service) and finally
make a call path between gateways and PSTN (Public Switched Telephone Network).
It manages the bandwidth. Allocate a certain amount of bandwidth for a call and select codec to use. It acts as a
regulator of bandwidth to protect the network against bottlenecks (congestion).
The gatekeeper is responsible of routing function. It must redirect calls to right person by the proper bridge.
Also, is able to handle many other functions such as conference call.
Finally, it can supervise several gateways. The gatekeeper, by its features routing and security, is able to manage
gateways to ensure that any appeal reaches its destination with the best quality service possible. It assures
redundancies bridges in order to bring about any call. It knows at any moment the status of each bridge and
route calls to accessible gateways.
Security is very important on a Gateway, it must be considered like a server. If your gateway is attack by DoS
(Denial of Service), it can alter the process of your VoIP network. Also, an intrusion can be dangerous for the
confidentiality of VoIP communication.
Figure 3.5. Exhibit of a Gateway at the customer Side
1.6.1. Comparative of Cisco Gateway Hardware
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Table 3.1. Cisco AS5300
Processor Type
150-MHz R4700
Memory
64 MB DRAM
Ethernet (RJ-45)
Two (one 10 MB, one 10/100 MB)
Wan Interface Options
Quad T1/PRI (RJ-45); Quad E1/PRI (RJ-45)
Table 3.2. Cisco AS5300
Processor Type
250-MHz RISC processor
Memory
256-MB (default) to 512-MB (maximum)
Ethernet (RJ-45)
Two (one 10 MB, one 10/100 MB)
Wan Interface Options
Eight-port CT1, CE1, PRI termination
Table 3.3. Cisco AS5850
Processor Type
650-MHz Broadcom 1250 dual core RISC processor
Memory
1.0-GB synchronous dynamic RAM (SDRAM) with
Error Correction Code (ECC)
Ethernet (RJ-45)
Dual gigabit load-balanced redundant Ethernet ports
with gigabit interface controller (GBIC) interfaces for
user traffic; Dual 10/100-Mbps Ethernet port with RJ45 connector for management traffic or call agent or
softswitch control traffic
Wan Interface Options
One CT3 and 216 DSP feature boards; 24 CE1/CT1
feature boards; One STM-1 feature board
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Chapter 4. Communication between
SIP and Asterisk
Course Objectives:
Know the SIP protocol funtionning
Install Asterisk
Configure some options
1. Course
1.1. Definitions
Dialog: Exchange between two User Agents for a given time. A dialog is a group of transactions.
Caller: The entity that initiate a session with an INVITE request.
Invitation: INVITE request.
Callee: The receiver of an INVITE request.
Message: Request or response exchanged by SIP elements.
Method: Indicates the type of request sent to a server. For example, the INVITE and BYE requests.
UAC (User Agent Client): An UAC is a logical entity that acts as the client in a client/server application. It’s
in charge of sending requests and receiving responses.
UAS (User Agent Server): An UAS is a logical entity that acts as the server in a client/server application.
It’s in charge of receiving requests and sending responses.
URI (Uniform Resource Identifier): An URI identifies an entity by using syntax, similar to the one used for
emails, in this form « sip:identifier@domain » (for example sip:john@sip.labo-voip.com).
Proxy Server: Intermediate entity, at the same time client and server, which provides routing service to
clients that try to reach other clients. Consequently, the Proxy server does requests by the name of other
clients.
Redirect Server: UAS that redirects to a set of alternative URIs by generating 3xx responses to requests it
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receives.
Registrar Server: Server that accepts REGISTER requests it receives and stores the information. It’s used to
identify/authenticate users.
Request: Sent by a client to a server, this SIP message permits to call upon a particular operation.
Response: Sent by a server to a client, this SIP message indicates the status of a previously sent request by a
client to the server.
Session: Multimedia flow exchanged between a set of callers and callees.
Transaction: Is composed of all the exchanged messages between a client and a server, from the first request
to the final response.
Stateful Proxy: Maintains the state for transactions between client and server.
Stateless Proxy: Transmits each request and response it receives without maintaining the state of the
transaction.
1.2. SIP functionality
1.2.1. User Agents
They are logical entities that use SIP to find another destination entity.
The User Agents can be (non exhaustive list):
Softphones (software applications)
IP phones (wired or Wi-Fi)
Smartphones and PDAs
IPBX
IP/TDM gateways
Figure 4.1. Simple example of a basic SIP architecture
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Each User Agent has an UAS and an UAC. It’s these logical entities that permit to send responses and receive
requests for the first one, send requests and receive responses for the second one. It’s important to remember
that the client or server state is only for the transaction duration. Thus, a User Agent is in turn client and server.
1.2.2. Proxy server
Important piece of the SIP architecture, it provides routing service to messages sent by a client, and by
maintaining certain important functions like:
The actual location of the callee
The accounting (for billing)
Etc.
The messages can travel across a set of Proxy servers, until reach the one who knows the callee location.
Figure 4.2. Transfer between Proxy servers to reach the destination
It exist two types of Proxy server explained just below:
Stateless Server
Simple and quicker than the Stateful server, it transmits messages independently from the others without
keeping in mind the state of the transaction. This fact, the Stateless Server doesn’t provide message
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retransmission mechanisms. However, it’s used for load balancing, message translation and routing.
Basically, a Stateless Proxy only forward messages as it receives then. So, it will not generate its own
temporary response messages for example.
Stateful Server
Contrary to Stateless Server, it maintains the state of the transaction, from the first request to the final
response. This feature includes an additional process time that make it slower, but offers very advantageous
functions:
The forking is a good example; it permits to redistribute a request to multiple destinations (session
initiation with multiple callees).
The message retransmission, because it knows the content of the transaction.
The user’s location, it’s then possible to redirect a call to he cellular phone of a user when the call was
initially transmitted to the office telephone.
The accounting.
Some help to NAT translation.
In general, the users of VoIP/ToIP networks use the corporation domain name for the network part of the URI.
The SIP Proxy servers are then identified via SRV type DNS entries, just like email servers are identified by
MX type entries. This permits to have a single URI, whatever the corporate SIP proxy in use.
The SRV type DNS entries are written by following the RFC 2782 form:
{_Service. Protocol} SRV {Priority} {Port} {Server name/IP}
Entries for a corporate DNS domain may look like this:
_sip._udp SRV 0 5060 sip.a.com
_sip._udp SRV 1 5060 backupsip.a.com
Thus, the entire user’s URI for this network may be written like this:
Without SRV type DNS entry
With SRV type DNS entry
user@sip.a.com
user@a.com
1.2.3. Registrar server
It’s a server that provides a way to localize the users. For that, the users register themselves by sending register
requests (REGISTER) to the server. This one extracts information about the current user’s location, like IP
address, port number and username. Then, it stores this information on a database.
Figure 4.3. Simple registration scheme on a SIP Registrar server
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The Registrar server can accomplish a simple identification, minimum process to localize the users on the IP
network. It’s also possible to implement authentication, to control the users connecting to the VoIP network.
It’s possible to identify or authenticate the caller and/or the callee.
1.2.4. Redirect server
The Redirect server permits to retrieve a list of current locations of a specific user. The database created by a
Registrar server is the source of information used by the Redirect server to create this list, which is sent on a 3xx
class of response. By this way, the caller can have a list of possible locations of the callee.
Figure 4.4. Simple redirect scheme on a SIP Redirect server
1.2.5. Other types of SIP servers
It exist multiple SIP servers, each one answering to a specific feature. We can list the following servers (non
exhaustive list):
Conference server
This server will proceed to RTP flow mixing coming from different participants, and provides functions
related to these conferences management.
Voice Mail server
It centralize vocal message functions. The vocal messages can then be managed via a vocal interface, a
HTML interface, or through the use of emails when a link is configured between this Voice Mail server and a
corporate email server (Microsoft Exchange for example).
IVR (Interactive Voice Response) server
The interactive voice servers permit to create vocal menus to process the calls. This kind of service is
especially used by help desks.
All the SIP servers can be separate network entities, or even mutualized in a single computer. It’s also possible
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to multiplicate some servers, for different reasons going from redundancy to load balancing.
1.3. SIP methods
1.3.1. SIP messages
The SIP communications are done through the use of a messages’ series that can be of two types:
Requests
Permits to call upon a particular operation.
Responses
Permits to inform the caller that his request has been received, processed, and even the result of this process.
Each message is composed by a first line indicating the type of message, the message header (SIP header) and
optionally a message body. The two last ones are separated by an empty line.
The message body can be of multiple types. The most common is a SDP message included into an INVITE
request.
The great flexibility of the SIP protocol comes from the liberty to create personalized requests and/or responses.
It’s then possible to create additional services.
1.3.2. SIP header
The SIP header is described by a list of fields. Here are the main ones:
Table 4.1. Main fields of the SIP header
Fields
Description
Via
Indicate the path taken by the message (typically the UAC address that has just sent
the message)
From
To
Indicates the source of the message
Indicates the destination of the message
Contact
Provides the URIs to reach the caller for future communications
Call-ID
Unique identifier that permits to distinguish a communication
CSeq
Content-Type
User-Agent
Content-Length
(Command Sequence) Unique transaction identifier for a specific session
Indicates the type of media of the message body
Character string describing the terminal used to send this message
Indicates the size of the message body
Here is an example of INVITE message sent:
INVITE sip:luc@sip.b.com SIP/2.0
Via: SIP/2.0/UDP 10.1.16.170:5060;rport;branch=C4BF7BAD282A1EA948DFA
From: John <sip:john@sip.a.com>;tag=3580587940
To: <sip:luc@sip.b.com>
Contact: <sip:john@10.1.16.170:5060>
Call-ID: FC9C664C-8134-47F2-877B-2ACBF60DB1B9@10.1.16.170
CSeq: 47647 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105x
Content-Length: 254
1.3.3. SDP header
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The SDP message, as the SIP message body, contains multiple fields classified in three categories:
Session description
Temporal description
Media description
It exist 20 different fields classified in the three categories from above. It’s useless to introduce all of them, but
it’s quite interesting to know the main ones:
Table 4.2. Main fields of the SDP header
Fields
Meaning
v
Version
SDP protocol version (v=0)
Description
o
Origin
Provides information about session origin (<username>
<session id> <version> <network type> <address type>
<address>)
c
Connection Data
Indicates connection data (<network type> <address type>
<connection address>)
t
Times
Provides information about session times (<start time> <stop
time>)
m
Media Announcements
Specifies transport details of flows on the network, the last
parameter indicating used codecs (described by « a=rtpmap »
fields) (<media> <port> <transport> <fmt list>)
a
Attributes
Different session attributes, serving here to enumerate the
different codecs that can be used for the communication
(rtpmap:<payload type> <encoding name>/<clock rate>)
RTP/AVP = Real-Time Transport Protocol using the Audio/Video profile carried over UDP
Here is an example of SDP header sent in an INVITE message:
v=0
o=john 16742548 16742652 IN IP4 10.1.16.170
s=X-Lite
c=IN IP4 10.1.16.170
t=0 0
m=audio 8000 RTP/AVP 3 98 97 101
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
1.3.4. SIP requests
It exist multiple SIP requests. Nevertheless, the most important ones are described just below:
INVITE
Permits to initiate a multimedia session.
REGISTER
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Contains information about the current location of the user, his IP address and port number. This request is
sent to a Registrar server.
BYE
Terminates an established session.
ACK
Acknowledges the final response of an INVITE request. The establishment time of a session, using a three
steps method, is random. Indeed, it depends on the time that a callee takes to accept or reject the call. Thus,
the callee periodically resends the response until receipt of the acknowledgement.
CANCEL
Cancel the session when being established. For example, when the callee take too much time to provide a
response.
1.3.5. SIP responses
The responses are identified by a code defined in the version 2 of the SIP protocol. The code can have a value
from 100 to 699, these values being classified in 6 categories:
1xx provisional response: The request processing can be quick or long. Thus, the 1xx responses inform the
caller that the request has been received and is currently being processed. This avoids the caller to resend the
request. The number 100 (TRYING) is used after INVITE requests, and the number 180 (RINGING) to
indicate it’s ringing on the other side.
2xx final positive responses: Indicates that a request has been processed and accepted. 200 (OK) is the final
positive response to an INVITE request for example.
3xx redirection: When a Proxy server can’t satisfy a call, it redirects the caller to an alternative service that
will be able to establish the call. This service can be another Proxy server or the new user’s location.
4xx final negative response (client error): Indicates that a request can’t be processed or the request uses a
wrong syntax and the problem comes from the caller.
5xx final negative response (server error): Indicates that the server can’t process the request although it’s
valid. The caller will send again the request later.
6xx final negative response (global failure): Indicates that the request can’t be processed by any server. In
general, the callee declines its participation to a session with a 603 response.
The first line contains a message in human language to explain the reason of the transmitted response by the
destination User Agent.
1.3.6. List of predefined SIP responses
It exist multiple predefined responses. The codes and their meaning are presented in this table:
Table 4.3. SIP responses
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Code (Message)
Meaning
100
Trying
180
Ringing
181
Call Is Being Forwarded
182
Queued
183
Session Progress
200
OK
202
Accepted
300
Multiple Choices
301
Moved Permanently
302
Moved Temporarily
305
Use Proxy
380
Alternative Service
400
Bad Request
401
Unauthorized
402
Payment Required
403
Forbidden
404
Not Found
405
Method Not Allowed
406
Not Acceptable
407
Proxy Authentication Required
408
Request Timeout
410
Gone
412
Conditional Request Failed
413
Request Entity Too Large
414
Request-URI Too Long
415
Unsupported Media Type
416
Unsupported URI Scheme
420
Bad Extension
421
Extension Required
422
Session Interval Too Small
423
Interval Too Brief
429
Provide Referrer Identity
480
Temporarily Unavailable
481
Call/Transaction Does Not Exist
482
Loop Detected
483
Too Many Hops
484
Address Incomplete
485
Ambiguous
486
Busy Here
487
Request Terminated
488
Not Acceptable Here
489
Bad Event
491
Request Pending
493
Undecipherable
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Code (Message)
Meaning
494
Security Agreement Required
500
Server Internal Error
501
Not Implemented
502
Bad Gateway
503
Service Unavailable
504
Server Time-out
505
Version Not Supported
513
Message Too Large
580
Precondition Failure
600
Busy Everywhere
603
Decline
604
Does Not Exist Anywhere
606
Not Acceptable
1.4. Description of an SIP session
1.4.1. SIP transactions
SIP is a transactional protocol; this implies that a request and all the associated responses have to be grouped in
transactions.
The transactions are easily identified, because all the SIP messages will use the same sequence number (CSeq).
However, there is an exception with ACK. Indeed, the ACK is not considered to be part of the transaction when
receiving a final positive response to a request because, even if there is only one request, multiple participants
can answer positively to it. On the other hand, the ACK is considered part of the transaction when receiving a
final negative response.
Figure 4.5. Transaction for call establishment
Figure 4.6. Transaction for call termination
1.4.2. SIP dialogs
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A SIP Dialog is an exchange of transactions between two User Agents in time. In addition, it eases scheduling
and routing of messages between SIP endpoints.
From a pragmatic point of view, a dialog is a logical suit of transactions.
The following fields of a SIP message permit to identify a dialog:
Call Id: Identifies a call composed of one or multiple dialogs. It also permits to distinguish the dialogs.
From: Identifies the dialog from the caller side.
To: The opposite, by identifying the dialog from the callee side.
CSeq: Orders the messages into the dialog and permits to identify a transaction.
Indeed, a dialog, and then the corresponding transactions, is composed of messages that chare the same
identification parameters. The dialog identification permits to two User Agents to keep their relationship by
using a Proxy server only once, when the two peers know their location.
Moreover, certain messages establish a dialog, others not. The best example is a BYE request, which takes place
in a pre-established dialog with an INVITE request.
Figure 4.7. Example of dialog cut in two transactions
1.4.3. Register
The registration of a client onto a SIP Registrar server is done via a REGISTER request. The server can be
configured for a simple identification, to pick up information about the location of the client, or for
authentication, in order to ensure the identity of the client.
Figure 4.8. Registration with identification
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Figure 4.9. Registration with authentication
1.4.4. Invite
The way INVITE requests are processed depends on the type of Proxy server used. A Stateless Proxy will only
redirect received messages to a destination, whereas a Stateful Proxy will be able to keep state of transactions,
and then to generate its own responses.
A Stateful Proxy can activate the Record Routing, in order to force all the following messages to travel across it.
For this, the Proxy adds a « Record-Route » field in the SIP header for all the requests sent.
Each Proxy using the Record Routing will add its « Record-Route » field.
Figure 4.10. Call establishment with a Stateful Proxy
Figure 4.11. Call establishment with a Stateless Proxy
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1.4.5. Session termination
The call termination is directly done between the two User Agents, unless if Record Routing is activated by the
Proxy. In this case, all the messages will go through the Proxy.
Figure 4.12. Call termination (without Record Routing)
Figure 4.13. Call termination (with Record Routing)
1.4.6. Example
This example is composing of 2 terminals, and 2 proxies. The User 1 want call the User 2, and will terminate it
later. The proxy A is a stateful proxy with record routing and authentication, and the proxy B is a stateless
proxy without record routing.
Figure 4.14. Example
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1.5. Asterisk
1.5.1. Introduction to TrixBox
Trixbox is the world's most popular Asterisk-based distribution. Even if you are novice user you can use
configure quicky a VoIP System to allow your IP phone and Softphone to work in your network
TrixBox is a package combining all the required software to install and use of an IPBX:
Operating System: GNU/Linux CentOS
IPBX: Asterisk
Database: MySQL
Administration interface: FreePBX
The TrixBox is use in small and medium company to provide a powerful solution of VoIP.
1.5.2. Installation
1.5.2.1. Installation in a virtual machine
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The following process has been writing on the version 2.6.1.13 of Trixbox, and using on Microsoft Virtual PC
2007. The ISO image can be directly downloaded here: \\labs\LABS\CISCO\VoIP or at
http://www.trixbox.org/downloads.
This is the first screen you will see, so to begin the installation press the key <ENTER>.
Now, you need to select your keyboard type, for example for QWERTY you can select “us”. After press the
key <TAB>, and you can press <ENTER>.
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In this part, you must select your time zone.
Next, you have to choose a passport for the root user (during a practice,we recommend you to use “password”
as the root’ password).
Now, it is the time of installation, wait it will take 5 minutes... (It depends of your hardware.)
When the Trixbox will restart, you must release the ISO (and in a real case, eject the CD-ROM). In a case of
virtualization check you have select the good network card interfaces (it depend if you use WiFi or Ethernet).
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This screen show you what is happen during the start. It show you several things, make attention in the part of
eth0, it represent your network card. If you have failed, that mean you don’t have select the good network card
in your virtualization software.
It is start now; you must enter the login “root” and the password you have chosen before. It shows you the URL
you can use to connect to the HTML interface (here: http://192.168.0.141).
If you have some trouble with your network you can use the command “ifconfig” to show you network
parameters, and the command “dhclient” to initialize DHCP.
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For example, if you want modify the password of the main account, you can type “passwd-maint” to change it.
1.5.2.2. Upgrade the TrixBox
In this part, you will learn the help command and how to upgrade your TrixBox.
You can type the command “help-trixbox” to know about all command you can use.
You can use the command “yum –y update” to upgrade the TrixBox. This screen show you what is happen
during an upgrade: it download packages, and install them.
1.5.3. Configuration
1.5.3.1. Manage with the HTML Interface
The Trixbox provide a powerful HTML Interface, this one is very simple to use. You just need read and fill the
different fields. You can enable module, create user, configure the music on hold, create conferences, etc…
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1.5.3.2. Connect to the admin panel
To connect to this interface, you need to enter the URL of your TrixBox on your Web Browser (i.e.
http://192.168.0.140 in the example), after you will have the following screen:
Now click on the link “switch”, the following screen appear. If you have never configured an administrator
account you can use the main account (login “maint”).
The following screen appears:
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Presentation of the menu
1.
System Status: This menu is a report your system status, for example network usage, memory usage, how
many users are online, etc...
2.
Packages: This menu concerns all packages you can use on your systems. For example if you want make a
PERL script to automate jobs on the server...
3.
PBX: In this menu you configure your PBX (User configuration, conferences, module enabled, etc...).
4.
System: This menu provides you more details of your system.
5.
Settings: This menu is dedicate to the TrixBox configuration; Which SMTP server you will use, which
reposities you will use for upgrade, or the information about your registration for subscribe to the official
support desk.
6.
Help: In this menu, you will find the official manual and support of the TrixBox.
When you click on PBX, then PBX Settings from the admin menu, you will have this following screen:
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As you can see you have the menu at the left, you can configure all part of your PBX.
Note
Be careful, after any modification, you must click on “Apply Configuration Changes” to be able to see
your alteration.
1.5.3.3. Introduce of Asterisk CLI
Also on a PBX, you can manage it with command line we call it Asterisk CLI. On a TrixBox to access it, you
must enter the command “asterisk –r” on your server. The Asterisk CLI is very similar than a Cisco IOS, for
example you can use TAB completion and the key “?” or you can use the command “help” to see all command
and their descriptions.
1.5.3.4. Extensions
An Extension is a logical representation of a user account. First step to create a new user, you need to click on
“Extension” from the PBX menu. After, you must select the type of your device (for example “Generic SIP
Device”).
Finally, fill the required fields:
User Extension
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This number corresponds to your telephone number, as well as your identifier to authenticate against the SIP
Registrar server.
Display Name
It’s the alphanumeric string that is displayed on the user’s terminal.
Secret
This alphanumeric string is the password used to authenticate against the SIP Registrar server (it’s
recommended to use only digital numbers to provide full compatibly with old devices).
Also, you can fill optional fields like the email address.
1.5.3.5. Conference
A conference is a logical room, who allow several users to be in the same discussion. First step to create a
conference, you must click on “Conferences” from the PBX menu.
After, fill the required fields:
Conference Number
This number corresponds to the dial number for this conference.
Conference Name
It’s the alphanumeric string that is displayed on the user’s terminal.
You can fill optional fields:
User PIN
This is password who users need to enter to access to this conference.
Admin PIN
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This is password that users need to manage this conference.
Allow Menu
This option provide a menu when admin press the key “*” (i.e. kick a user; mute a user; etc…).
1.5.3.6. Voice Mail
The voice mail allows you to receive a voice message in your mail box. When you are absent, your callers can
be allowed to let you a voice message.
To setup it, you need to select a user extension, scroll down until the section “Voicemail & Directory” and fill
the following fields:
Status
The statement of the voicemail for this user.
Voicemail password
This is password the user need to enter to access to the voicemail interface.
Email Address
The email address where is sent the notification of a new message.
Email Attachment
This option provides the voice message attached to the email sent.
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In VoIP context a voicemail interface is called “ARI Interface”. To access it in the Trixbox your need to go at
the
following
URL
and
enter
the
login
and
the
password
of
the
user:
http://192.168.0.141/user/index.php?vmrecs (replace “192.168.0.141” by the IP address or his domain
name of your server.).
1.5.3.7. Ring group
A ring group is a group of extension with a ring policy. For example, you want allow user to call the sales
services, you can make a group with all extension of this services. The particularity of this group is you can set a
strategy on the ring; call everyone, call the first available, etc…
To configure it, you need to select “Ring groups” from the PBX Menu, and fill the following fields:
Ring group Number
The dial number of your group.
Ring Strategy
The ring policy you want use for this group.
Extension List
The lists of extensions are member of this group.
Destination of no answer
Is the destination use if nobody answers.
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1.5.3.8. Queue
A queue is a FIFO (First In First Out) list of caller. That mean, if a new caller enter in the queue, he need to wait
people who enter before him be processed.
To setup it, you need to select “Queue” from the PBX Menu, and fill the following fields:
Queue Number
The dial number of your queue.
Queue Name
The name of your queue.
Static Agents
The lists of extensions who are able to answer to this queue.
You can fill optional “Caller Position Announcements” fields to announce the user position.
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1.5.3.9. IVR
An IVR (Interactive Voice Response systems) automates interactions with callers. It used pre-recorded voice
prompts to make menu to help users to select the category who correspond to their needs. More and more this
technology is used by company to provide a powerful hotline. This one is able to manage queue and redirect to
the good. When you want create an IVR, you must imagine first all points (the start to the end), because you use
reverse engineering to configure it. That mean, to be able to setup one IVR, you must start by end points, and
finish by the start point.
We want for example creates this IVR:
The first step we need to create is the end points. The order you must use to setup this IVR is:
7.
Create the Voice mail for the Billings services
8.
Create the Voice mail for the Sales services
9.
Create the Queue for the Support services
10.
Create the Rings Group for the Sales services
11.
Check or create the extension 7102.
12.
Create the announce message FAQ.
13.
Create the announce message Start.
14.
Setup on your IVR user can access to the directory server with the key #.
15.
Create your IVR.
16.
Setup the inbound routes to access to your IVR.
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1.5.3.10. Trunks
A Trunk is interfaces that allow make an interconnection between your TrixBox, POTS (Plain Old Telephone
Service) equipment, and VoIP network. In the TrixBox you can setup several type of Trunk. For example, if you
configure an SIP Trunk. This one allows your Trixbox to be interconnected to another SIP networks. Also, to
define witch way will be use by your users, you must configure dial rules (i.e. all number begin by a “0”).
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Chapter 5. Communication with
SCCP and Call Manager Express
Course Objectives:
Understand the aim and the mechanis of CCME
Know how to configure CCME
1. Course
1.1. Generalities
1.1.1. Introduction
SCCP is a Cisco proprietary protocol used for real-time communications and conferences.
The benefits of SCCP protocol based on its low memory requirements and CPU load. This protocol can be used
in a secure LAN with a qualify bandwidth sufficient.
One disadvantage of SCCP is the management of the QOS and bandwidth. Likewise, the PRTC protocol
(Compressed Real-Time Transport Protocol) is not supported. SCCP doesn’t authenticate remote users outside
the CME LAN.
Despite, the use of VPN connection, SCCP remains unable to manage remote users. Each site must have a Cisco
CME to authenticate local IP phones. The operation through the WAN between several CME routers is done
through the H.323 protocol.
1.1.2. Call Establishment
The calls are centralized around the routers. When an incoming call arrives on a router, it is treated separately
until the destination is decided. As soon as the destination is known, an outgoing call is established. Then, the
connection between these two terminals is established.
Figure 5.1. This exhibit describes the logical way a connection is made between two
terminals.
1.1.3. Aim and Mechanism of CCME
Cisco Call Manager Express (CCME) is a call management solution based on Cisco routers that provide a
telephone services for about 300 users.
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Cisco CME is part of Cisco IP Communications solution and works in conjunction with Cisco System products,
including routers, switches, gateways, gatekeepers who translate a phone number into an IP address in the H.323
solution, a messaging service (Cisco Unity voice mail), ATA adapters (Analog Terminal Adapters), as well
access to public switched telephone network (PSTN : Public Switched Telephone Network).
Cisco CME support about 120 IP phones and offers lot of services and benefits of IP Telephony without the
high cost and complexity of deploying a solution based on servers.
The routers must first be equipped with IOS 12.3(7)T IP-Voice at least to manage the CME in the form of a
package to download on the router’s flash memory. The package includes, among the CME software, firmwares
for IP Phone and other files.
Figure 5.2. Cisco 1700, Cisco 2600XM, Cisco 3700
The CME system offers the PBX functionalities, and others dedicated on the IP Phones. All are centralized on
the Cisco CME router, who control all calls made and receive.
The IP Phones register themselves on the Cisco CME at startup, then they are able to receive and send calls. The
IP Phone and the CME communicate using the SCCP protocol (Skinny Client Control Protocol).
When a call is made from an IP Phone to another, it must go through a control phase of CME, the SCCP
protocol is used here. The SCCP protocol does not transfer from one phone to another directly, but between an
IP Phone and CME. Once the call is accepted, the protocol RTP (Realtime Transport Protocol) take over and
convert the voice into IP packets in UDP.
If Cisco CME needs to make a call to an IP phone managed by another CME, the H.323 protocol will be use to
make the connection between the two CME.
The function of PSTN gateway (Public Switched Telephone Network) can be enabling on the CME router or
separate bridges. In this case, the IP-to-IP function will be activated to enable the translation between H.323
protocol and SIP.
1.2. Cisco IP Phone
1.2.1. Topologies
There are 3 methods to install a VoIP infrastructure in the company network :
Figure 5.3. Single cable
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This method is the best, because there is only one cable per user. The problem is you must to have Cisco
switches, or switches with Voice capability. You just have to configure VLANs and Trunk protocol.
Figure 5.4. Multiple
This second method is good, but you must install 2 cables per users. The installation is more expensive and you
might have to buy and install others switches. It’s easy to configure VLAN and Trunk to separate the range
ports for Voice Network and other range for the Data Network.
Figure 5.5. Multiple Switch
For this method, the infrastructure is completely doubled, but the cost of the install is very expensive. However
you have two networks, which is more secure. If your Data Backbone and your Telephony backbone are
physically remote, it is more easy to install your Voice Network.
1.2.2. Register Steps
This process is divided into steps:
Step 1: The switch send a special tone called « Fast Link Pulse » (FLP) from his interface. The FLP will be
forwarded to the Powered Device (PD) in this case represented by an Ip Phone.
Step 2: When the Powered Device is not supplied, it creates a link between the incoming interface and
outgoing interface, creating a loop. So this loop can refer the FLP to the switch. In the end, if the FLP not
return the switch, no power will be sent on this interface.
Step 3: After the return of FLP, the switch will sent the power on this interface.
Step 4: The line is activated within 5 seconds “link up”.
Step 5: The IP Phone startup.
Step 6: With the Cisco Discovery Protocol (CDP), the IP Phone announce at the switch the amount of power
is needed.
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Step 7: With again CDP, the switch inform the IP Phone the list of available Voice VLAN “Auxiliary
VLAN”.
Step 8: The IP Phone registers with a DHCP server with DHCP-Discover request on broadcast to obtain an IP
address in the VLAN voice pool.
Step 9: The DHCP server send all IP parameters to the IP Phone. The IP address of TFTP server is the CME
IP.
Step 10: The IP Phone apply the configuration.
Step 11: The IP Phone connect to the TFTP server and download the XML configuration file
«SEP00112FD21239.cnf.xml » (00112FD21239 represent the MAC address of IP Phone).
This file contains the register information’s for the Cisco CME, IP address, the language, the port and the
firmware. If the IP Phone have the good firmware, is register and receive the configuration.
Note, that the XML SEP file does not contain the extension number.
Step 12: If the firmware is obsolete or different, the IP Phone will download the new firmware from the
TFTP server.
Step 13: The IP Phone reboot after the firmware download.
Step 14: If no SEP XML file exist with MAC address of device, is that a new added IP Phone. The IP phone
will download from TFTP server a file named XMLDefault.cnf.xml who indicates IP Address, port number
and the firmware to use by the IP Phone. The process is the same as above: Download Firmware, and reboot
if necessary.
Step 15: The IP Phone will register at the Cisco CME using SCCP messages type. If “auto assign” option is
activate, the IP Phone will receive automatically an extension from Cisco CME. If “auto assign” option is not
activate, the IP phone has no extension and will be unable to send and receive calls
1.3. Switches configuration
1.3.1. The flow separation
One of the bases about VoIP is the voice and data transmission on the same media. The flow separation will be
done by VLAN: One Voice VLAN and One Data VLAN. Don’t forget, the VLAN1 is used for the management,
so you must create 2 VLANs on each router.
The CISCO IP Phones can be considered as Layer 3 switches, so they support the Trunking
The 3 ports of IP Phone allow:
A connection 10/100 on Ethernet to the switch,
A connection 10/100 on Ethernet to the computer,
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An internal port for the Audio stream.
The 10/100 Ethernet port connected on the switch send the 802.1Q protocol (trunking). That permit to connect
the IP Phone at the Voice VLAN (auxiliary VLAN) and connect Computer at the Data VLAN.
The 1st pattern represent an IP Phone connected on one side to one switch, and the other side on the computer.
All IP addresses are on the same subnet.
The 2nd pattern represents the different subnet for the Computer and IP Phone. In this case, it’s preferable to
used on the Voice VLAN the VoIP QoS different than a classical LAN.
Other architecture is possible. It’s based on each equipments (Computer and IP Phone) are directly connected on
the switch. The problem is the wires number. Because, we need 2 more cables.
Packets from IP Phone pass on the other VLAN than others packets. This separation allows simplify the
deployment process, because, when a new IP Phone as connected, he is automatically configured on the good
VLAN. The IP Phones and switches communicate with CDP protocol. So on the startup the switch give the
configuration at the IP Phone:
VLAN ID (VVID),
Port VLAN ID (PVID).
Figure 5.6. One port is associated at 2 VLAN: Voice for IP Phone and Native for
Computer.
1.3.2. Connect IP Phone at the Network
Two methods exist to connect IP Phone on the Network:
With a Single cable:
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Lot of Company used this architecture, because they don’t need to setup more cables, more switches… so is
less expensive.
With two cables:
This architecture is based on the cables separation. It’s easier to configure the priority (QoS), but is more
expensive.
1.3.3. VLAN configuration
Two VLAN will be necessary for the functioning of the IP phone and Computer. Computer is connected at the
DATA VLAN, and IP Phone is connected at the VOICE VLAN. The trunking will be required to allow the flow
of information of these two VLAN, between the IP phone and the switch
The commands are identical to thoses of a typical VLAN configuration on the Catalyst switch, with the
exception of creating a Voice VLAN
Switch(config-if)#switchport voice vlan {number}
Interface configuration Mode
Assign one port of the voice vlan.
Console(config)#interface FastEthernet0/1
Console(config-if)#switchport trunk encapsulation dot1q
Console(config-if)#switchport trunk native vlan 1
Console(config-if)#switchport access vlan 12
Console(config-if)#switchport mode trunk
Console(config-if)#switchport voice vlan 112
Console(config-if)#spanning-tree portfast
The “Access VLAN” is used between the Computer and the LAN. The Voice VLAN is used by the IP Phone for
communicate an audio stream. Don’t forget the native VLAN, BLAN 1, where the positions of Computers have
not received a specific VLAN.
Switch# show interface fa0/17 switchport
Name: Fa0/17
Switchport: Enabled
Administrative mode: trunk
Operational Mode: trunk
Administrative Trunking Encapsulation: dot1q
Negotiation of Trunking: Disabled
Access Mode VLAN: 0 ((Inactive))
Trunking Native Mode VLAN: 12 (VLAN0012)
Trunking VLANs Enabled: ALL
Trunking VLANs Active: 1-3,5,10,12
Pruning VLANs Enabled: 2-1001
Priority for untagged frames: 0
Override vlan tag priority: FALSE
Voice VLAN: 122
Appliance trust: none
The inter-VLAN routing done by the Layer 3, and requires a router to make this connection. IP phones are
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presents on the Voice VLAN and Computers on the Data VLAN. The Trunk used on the port connected at the
switch.
1.3.4. DHCP configuration
Router(dhcp-config)# option {option-number} ip {IP-address}
DHCP configuration Mode
Set the specific value option of DHCP
Router(config)#ip dhcp exluded-address 10.90.0.1 10.90.0.10
Router(config)#ip dhcp pool mypool
Router(dhcp-config)#network 10.90.0.0 255.255.255.0
Router(dhcp-config)#option 150 ip 10.90.0.1
Router(dhcp-config)#default-router 10.90.0.1
Router(dhcp-config)#dns-server 10.100.0.1 10.100.0.2
Router(dhcp-config)#exit
Option 150 is the IP address of TFTP server. In this case is the IP address of CME router. Don’t forget to
exclude the IP range of router (dhcp excluded-address), to indicate a default gateway (default-router) and
DNS server (dns-server), one a minimum.
1.3.5. Routers configuration
1.3.5.1. Ephone and CME configuration
Router(config)# ephone {phone-tag}
Global Configuration
Create an ephone
Router(config-ephone)# mac-adress {mac-adress}
Ephone configuration mode
Associate MAC address at the ephone
Router(config-ephone)# button {button-number} {separator} {dn-tag}
Ephone configuration Mode
Assign a number to call ephone-dn to a button on the ephone
{separator} is a unique character that defined the properties button and extension number:
« : » : Standard ringtone.
« b » : The tone is off, but the beep of call waiting is allowed.
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« f » : Ring, to differentiate calls for a line over another. The Ring is three pulses instead of a pulse for
calls and two calls for external.
« m » : Monitor mode for the sahred-line wich indicates which lines are used or not.
« o » : Several lines ephone-dn share a button (10 lines max). The field dn-tag contains dn-tag separated by
comas.
« s » : Ringtone silent, only the icon (flashes on the phone to report a call).
This command can be repeat several times : Router(config-ephone)# button {button-number} {separator} {dntag} {button-number} {separator} {dn-tag}
Router(config-ephone)# type {7940 | 7960} addon 1 7914 [2 7914]
Ephone configuration Mode
Configure one or several modules 7914 for ephone type 7940 or 7960.This command is obligatory if we use
the extension module 7914.
1.3.5.2. Example of basic configuration
CMERouter(Config)#ephone-dn 7
CMERouter(Config-ephone-dn)#number 1001
CMERouter(Config)#ephone 1
CMERouter(config-ephone)#mac-address 000F.2470.F8F8
CMERouter(config-ephone)#button 1:7
One ephone-dn 7 is created and associated at the Ephone 1. The ephone-dn is configured on single-line and is
attributed at the button 1 on the IP phone with MAC address associated.
Each Ephone will be one or several ephone-dn assigned at the buttons on the physical equipment.
1.3.5.3. Example of configuration with several Ephone
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CMERouter(config)#ephone-dn 10 dual-line
CMERouter(config-ephone-dn)#number 1004
CMERouter(config)#ephone-dn 11 dua-line
CMERouter(config-ephone-dn)#number 1005
CMERouter(config)#ephone-dn 12 dua-line
CMERouter(config-ephone-dn)#number 1006
CMERouter(config)#ephone-dn 13 dua-line
CMERouter(config-ephone-dn)#number 1007
CMERouter(config)#ephone 1
CMERouter(config-ephone)#mac-address 000F.2470.F8F1
CMERouter(config-ephone)#button 1:10
CMERouter(config)#ephone 2
CMERouter(config-ephone)#mac-address 000F.2470.A302
CMERouter(config-ephone)#button 1:11
CMERouter(config)#ephone 3
CMERouter(config-ephone)#mac-address 000F.2470.66F6
CMERouter(config-ephone)#button 1:12
CMERouter(config)#ephone 4
CMERouter(config-ephone)#mac-address 000F.2470.7B54
CMERouter(config-ephone)#type ata
CMERouter(config-ephone)#button 1:13
Each Ephone will be one or several ephone-dn assigned at the buttons on the physical equipment.
1.3.5.4. Example of configuration with several buttons on several Ephone
CMERouter(config)#ephone-dn 14 dual-line
CMERouter(config-ephone-dn)#number 1008
CMERouter(config)#ephone-dn 15 dual-line
CMERouter(config-ephone-dn)#number 1009
CMERouter(config)#ephone-dn 16 dual-line
CMERouter(config-ephone-dn)#number 1010
CMERouter(config)#ephone-dn 17 dual-line
CMERouter(config-ephone-dn)#number 1011
CMERouter(config)#ephone 5
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CMERouter(config-ephone)#mac-address 000F.2470.FAA1
CMERouter(config-ephone)#button 1:14 2:15
CMERouter(config)#ephone 6
CMERouter(config-ephone)#mac-address 000F.2470.A7E2
CMERouter(config-ephone)#button 1:16 6:17
In this example, there are two ephone configured with multiple lines on each. Each button is assigned to an
ephone-dn.
1.3.5.5. Phones Installation
You can used three methods for to install yours phones.
Router(config)# telephony-service
Global configuration mode
Enter on the Telephony configuration mode
Router(config-telephony-service)# max-ephones {number}
Telephony Configuration mode
The max number of phones is supported by the router.
Manual installation :
Router(config-telephony-service)# max-directory-numbers {number}
Telephony Configuration mode
The max number of the dns on the router
Router(config-telephony-service)# load {modele}-[modele] {firmware}
Telephony Configuration mode
The type of the IP phone used (7902, 7905, 7910, 7912, 7914, 7920, 7935, 7936, 7960-7940) with firmware
(ex: P00303020214).
Router(config-telephony-service)# ip source-address { ip-address} {port} [any-match/strict-match]
Telephony Configuration mode
The IP address of the Cisco CME router for authenticate the IP Phone. The default port number is 2000
Router(config-telephony-service)# create cnf-files
Telephony Configuration mode
Make the XML configuration files for the Clients
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Router(config-telephony-service)# keepalive {seconds}
Telephony Configuration mode
The interveal between 2 keepalive messages send by the router to the IP phone. The default time ris : 30sec.
The range value is : 10 – 65535.
Router(config-telephony-service)# user-locale {language-code}
Telephony Configuration mode
Configure the language on the IP Phone (US for English, FR for French…)
Router(config-telephony-service)# network-locale {language-code}
Telephony Configuration mode
Configure the tone of IP Phone (US for English, FR for French…)
Router(config-telephony-service)# reset {all [time-interval] | [cancel] | [mac-address] | [sequence-all]
Telephony Configuration mode
between each IP phone || cancel, stop the command || mac-address define the IP phone to make a reboot ||
sequence-all, and Each IP Phone start the reboot when the previous IP phone is restarted).
Automated phone installation:
You can execute a command to start a setup mode of Telephony. It’s same Principe that the Setup mode on the
first router boots.
The command is:
Router(config)# telephony-service setup
Global configuration mode
After this command, the router asked you, for example:
Do you want to setup DHCP service for your IP phones? [yes/no]
Do you want to start telephony-service setup? [yes/no]
Enter the IP source address for Cisco CallManager Express services :
Enter the Skinny Port for Cisco CallManager Express services [2000]:
How many IP phones do you want to configure [0]:
Do you want dual-line extensions assigned to phones? [yes/no]:
What language do you want on IP phones?
0 English
1 French
2 [...]
Which call progress tone set do you want on IP phones :
0 United States
1 France
2 Germany
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3 [...]
What is the first extension number you want to configure [0] :
Do you have Direct-Inward-Dial service for all your phones? [yes/no] :
Do you want to forward calls to a voice message service? [yes/no] :
Call forward No Answer Timeout : [18]
Do you wish to change any of the above information? [yes/no] :
Installation partially automated of IP Phone:
This configuration mode is a simplification of the manual mode; go into configuration mode telephone services
to enter this command:
Router (config-telephony-service)# auto assign [dn-tag] to [dn-tag] type [phone-type] cfw [extn-number] timeout
[seconds]
Telephony service configuration mode
1.3.5.6. GUI Configuration
In 1st, on the CME router, you must download some files for the GUI. The package is named: cme-gui3.1.1.zip. “3.1.1” represents the CME version, and you can download this package on the Cisco Website.
This archive contains:
XML Template:
xml.template
GUI files:
admin_user.html
admin_user.js
CiscoLogo.gif
Delete.gif
dom.js
downarrow.gif
ephone_admin.html
logohome.gif
normal_user.html
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normal_user.js
Plus.gif
sxiconad.gif
Tab.gif
telephony_service.html
uparrow.gif
xml-test.html
After, we can configure the GUI:
On the GUI, you can configure three users’ classes:
as the authorization to change parameters on the IP Phone.
The System administrator account must create by this command:
Router(config-telephony-service)# web admin system name {username} password {password}
Telephony service configuration mode
Make a username and password for System Administrator.
The option “secret 0 or secret 5” can be used for to crypt the password.
After, the system Administrator can log to the GUI interface with Internet Explorer or other. The Customer
Administrator and the Phone user can be create on the CLI, but it’s preferable to create on the GUI.
The command to create the Customer Admin is:
Router(config-telephony-service)# web admin customer name {username} password {password}
Telephony service configuration mode
The functionality of GUI for “Customer Administrator and Phone user” can be modifying on the XML files
(files with .css extension).
The creation of Ephone user doesn’t use on the same mode:
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Router(config)# ephone {phone tag}
Global configuration Mode
Permit to enter on the IP telephone configuration mode.
Router(config-ephone)# username {username} password {password}
IP telephone configuration mode.
Creation of the User Phone account.
GUI configuration :
Router(config)# ip http server
Global configuration Mode
Active the http server on the router
Router(config)# ip http path {flash :}
Global configuration Mode
Specify the location of the HTTP files.
Router(config)# ip http authentication {aaa | enable | local | tacacs}
Global configuration mode
Determines the authentication method on the http server.
After you can access on the GUI interface with this address: http://10.0.0.1/ccme.html (10.0.0.1 Represent the IP
address of the router).
On the GUI you can configure some operations:
On the System Administrator :
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Configure: ephone, ephone-dn, system parameters.
Voice Mail: Voice mail parameters.
Administrator: Make backup and restore, restart the CME router.
Reports: Shown some information’s or error messages.
Help: Link to the Help files.
If you want to create Customer and User account, you must go on the configure menu and choice system
parameters.
The rights of “customer administrator” are configured on the xml.template:
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After you have made modifications and download on the CME router. You must apply this with this
command:
Router(config-telephony-services)# web customize load {filename}
Example of result of this Customer interface:
Figure 5.7. In the left, the classical interface, and on the right, the interface with
modifications
For the Phone user account, it the same process, on the configuration page:
Phone user Interface :
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Configure: limited configuration of the phone user.
Search: Make a search on the Cisco CME Directory.
Help: Link to the Help files.
You can configure the call forwarding:
The Cisco CME Directory:
The System Administrator can configure the Directory presentation: (firstname, NAME or NAME, firstname)
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