Huma Mukhtar

Roll No. 02
Session 2018-22
Assigned By
Prof.Muhammad Usman
Why VoIP?

 Here are three reasons why VoIP is so

important:
Options
One big feature of VoIP telephony is that it offers many options to users. Users now have a
choice on many aspects of their phone usage. They can communicate through an audio-only
conversation, as well as a video conversation.
Hardware options
VoIP systems allow users to choose their hardware. Traditional phones, IP phones, computers and
smartphones can now be used to make calls, and so phone calls are no longer restricted to a
single device.
Convergence
With VoIP, the convergence of all forms of communication to the Internet is now complete:
emails, phone calls, video calls, conference calls and all other forms of data transfer can now
take place on a single unified IP network. This allows deployment and maintenance costs to be
reduced, while also removing the cost of additional networks, since the fees related to
expanding one’s Internet network to encompass telephony are much lower than the costs of
deploying and maintaining a separate network dedicated to phone calls.
Main Objectives

What Is VoIP ?
How Does VoIP Work ?
Step involved for originating a call
Basic Configurations
PC-to-PC architecture
 the phone-to phone architecture
VoIP Standard/Signaling
H.323 Standard
The H.323 Architecture
Protocol Relationships in H.323
The Seven Phases of an H.323 Call
Technical Challenges of VoIP
 What Is VoIP ?
 VoIP (voice over Internet Protocol) is the transmission of
voice and multimedia content over an internet connection.
VoIP allows users to make voice calls from a computer,
smartphone, other mobile devices, special VoIP phones and 
WebRTC-enabled browsers
 How Does VoIP Work ?
VoIP comprises several interconnected processes that
convert a analogue voice signal into digital format and a
stream of packets, transmit them on a packet-switched
network, and convert them back into voice at the destination

To accomplish this, VoIP will use . codecs

Codecs are either a hardware- or software-based
process that compresses and decompresses large
amounts of VoIP data. Voice quality may suffer when
compression is used, but compression reduces
bandwidth requirements..

https://voipstudio.com/blog/what-is-a-voip-codec-matter/
 Transmitting on packet switch
network

node
node

Converting voice
node1
into packet streams

Converting back
node
node into voice
Steps:
 The basic steps involved in originating an Internet telephone call are
conversion of the analog voice signal to digital format
 translation of the digital data into packets for transmission over the
Internet. The process is reversed at the receiving end
 The digital speech data is processed in units known as frames, with each
frame containing a portion of a speech signal of a specific duration.
 These frames are inserted into IP packets, which contain additional
information (overhead) such as packet sequence numbers, IP addresses,
and timestamps, all of which are necessary for the packet to traverse the
network successfully.
 To reduce the inefficiencies caused by this overhead, it is common to
pack several voice frames into one IP packet. Packetization of voice is
illustrated in Figure 2.18(a). The IP packets are received in a play out
buffer at the receiver, decoded in sequential order and played back
 Header Voice frame 1 Voice frame n

Packetization of voice

Audio
signal

Encoder
sender
packetsizer internet decoder

Dynamic Receiver
buffer
basic processes in VoIP.
Audio
signal
 Basic Configurations

The two basic models for VoIP

 PC-to-PC architecture
 the phone-to phone architecture
PC-to-PC architecture
 The PC-to-PC configuration is based on the assumption that users have access to multimedia
computers that are connected to the Internet. These computers can be on a LAN, as in a corporate
environment, or connected via telephone lines to Internet service providers (ISPs), as in home use.
 All encoding and packetization of the voice signal occurs in codec hardware and software on the
sender’s PC, while play out of the received signal occurs in the sound card on the receiver’s PC.
 Alternately, the codec could be implemented in hardware as part of a modem, network interface
card, or sound card.
 A user places a call by specifying the IP address of the recipient or by looking up the recipient’s
name in a public directory such as the dynamic host configuration protocol (DHCP) service provided
by ISPs
phone-to-phone architecture
In the phone-to-phone architecture, the user calls the Internet telephony
gateway that is located near a central office (CO) switch or a PABX. The gateway
prompts the user to enter the phone number of the recipient and initiates a VoIP
session with the gateway that is closest to the recipient. This gateway then places a
call to the recipient’s phone. End-to-end communication can proceed
subsequently, with voice sent in IP packets between the two gateways. Encoding
and packetization occurs at the sender’s gateway while decoding, reassembly, and
replay occur at the recipient’s gateway. The CO or PABX may digitize the message
or pass it on to the gateway for digitization. The signal on both users’ local loops is
analog. The gateways are implemented in hardware.
VoIP Standard
Signaling is one of the most important functions in
the telecommunications infrastructure because it
enables various network components to
communicate and interwork with each other.
 The H.323 Standard
H.323 is part of a family of ITU-T recommendations called H.32x, which provides
multimedia communication services over different networks. This family of standards is
summarized in Table

.
 describes protocols for the provision of audio-visual (A/V) communication sessions on
all packet networks
H.323 provides standards for equipment, computers and services for multimedia
communication across packet based networks and specifies transmission protocols for
real-time video, audio and data details.
 The H.323 Architecture
H.323 specifies four kinds of components, which, when networked together, provide
the point-to-point and point-to-multipoint communication services.

• terminals
• gateways (GWs)
• gatekeepers (GKs)
• multipoint control units (MCU

A typical H.323 network is composed of a number of zones interconnected via a

WAN.
 Each zone consists of a single H.323 GK, a number of H.323 terminal endpoints, a
number of H.323 GWs, and a number of MCUs interconnected via a LAN.The only
requirement is that each zone contains exactly one GK, which acts as the
administrator of the zone.
 Terminals
 An H.323 terminal, or client, can either be a personal computer or a stand-alone
device such as an IP phone, video phone, or a terminal adapter that connects an
analog phone or fax machine to the H.323 network.
It is an endpoint in the network that hat provides for real-time two-way
communications with another H.323 terminal, GW, or MCU.
This communication consists of control, indications, audio, video, or data
between the two terminals.
A terminal may set up a call to another terminal directly or with the help of a GK.
H.323 terminals may be used in multipoint conferences.

Gateways The gateway is an endpoint, which connects two dissimilar

networks.
 It provides translation of call signaling, media stream formats, multiplexing
techniques, and transferring information between the H.323 terminal and other ITU
terminal types for different types of networks.
For example, a gateway can connect and provide communication between an H.323
terminal and a PSTN phone
 . A gateway is not required for communication between two terminals on the
same H.323 network. A gateway may be able to support several simultaneous
calls from the LAN to several different types of networks

Gatekeepers The gatekeeper can be considered the brain of the H.323 network.
Although they are not required, gatekeepers provide functions such as
 addressing,
 authorization and authentication of terminals and gateways,
 bandwidth management
 accounting
 billing,
 charging.
 provide call-routing services.
 Clients register with the gatekeeper when they go online. One of its most
important functions, bandwidth management, is done through admission
control. Terminals must get permission from the gatekeeper to place or
accept a call. This permission includes a limit on the amount of bandwidth
the terminal may use on the network.
MCUs
MCUs support conferences of three or more H.323 terminals. All terminals
participating in a conference establish a connection with the MCU. The MCU manages
conference resources, negotiates between terminals for the purpose of determining
the codecs to use, and may handle the media stream
Protocol Relationships in H.323

H.323 is an umbrella for the following protocols :

When dealing with H.323, it is good to realize that it is not a
single protocol but rather an entire group of protocols. The
individual protocols used under the umbrella of H.323 include:
 H.225.0 for call signalling;
 Q.931, a protocol borrowed from ISDN, also used for call signalling;
 H.245 for negotiating audio/video channel parameters;
 H.235 for security and authentication;
 RTP, the Real Time Protocol , used to transmit audio/video streams;
 H.450.x for additional services like call transfer, call diversion, etc.
 The Seven Phases of an H.323 Call
Phase Protocol Functions

1. Call admission RAS RAS Request permission from GK to make/receive a call. At the
end of this phase, the calling endpoint receives the Q.931
transport address of the called endpoint.

1. Call setup Q.931 Set up a call between the two endpoints. At the end of this
phase, the calling endpoint receives the H.245 transport
address of the called endpoint

1. . Endpoint H.245 Negotiate capabilities between two endpoints. Determine

capability master-slave relationship. Open logical channels between two
endpoints. At the end of this phase, both endpoints know the
negotiation and RTP/RTCP address of each other
logical channel
setup

1. Stable call RTP/RTCP Two parties in conversation. RTP is the transport protocol for
packetized voice. RTCP is the counterpart that provides control
services.

1. Channel closing H.245 Close down the logical channels

1. Call teardown Q.931 Tear down the call.

1. Call disengage RAS Release the resources used for this cal
 THE GATEKEEPER-ROUTED CALL MODEL
We will assume a network that uses a gatekeeper and will also assume the
signalling flows via the

So, let's have two endpoints (IP phones) and a gatekeeper. The telephone numbers
assigned to the endpoints are 121 and 122, respectively. Let us assume the two
endpoints are registered with the gatekeeper.
Now, someone at the number 121 dials "122".
This is what is going to happen:
1. The endpoint that initiates the call knows the called number (122) but it does not know
the IP address associated with that number. At the same time, since it is registered with
the gatekeeper, it must ask the gatekeeper for a permission to place the call. It does so
by sending the Admission Request message to the gatekeeper. The Admission
Request (ARQ) will contain the called number (122), indicating to the gatekeeper that
the endpoint needs to have the number resolved to an IP address.
the protocol used to communicate with the gatekeeper is H.225.0-RAS
2. The gatekeeper will check it's database of registered endpoints whether it contains the number 122.
If so, the gatekeeper will check if 121 is allowed to call 122 and if it is possible to place the call — for
example, if there is enough bandwidth After that, the gatekeeper will form an answer — the
message Admission Confirm (ACF) with an IP address and send the ACF to the calling endpoint.
3. The enpoint 121 will now open a call signalling channel to the address provided by the
gatekeeper in the ACF message.
• With the gatekeeper-routed call model, the endpoint 121 will open a TCP channel to
the gatekeeper and send the Q.931/H.225.0 message Setup.
• The gatekeeper will in turn open a second TCP channel to the endpoint 122 and
forward the Setup message.
4. The endpoint 122 will first respond with the Q.931/H.225.0 message Call
Proceeding to indicate it has started working on setting up the call and the
gatekeeper will forward the message to the calling endpoint. After that, 122 will ask
the gatekeeper for a call permission (Admission Request, ARQ) and the gatekeeper
will respond with Admission Confirm (ACF). This is shown in Figure C.
5. The called telephone (122) starts ringing and this is signalled to the other party with
the Q.931/H.225.0 message Alerting, as shown in Figure D.
6. The called party picks up the handset and the endpoint can signal the call has been
accepted. This is done by sending the Q.931/H.225.0 message Connect.
At this point, the parties will need to negotiate parameters for audio (and optionally
video) channels. The protocol H.245 is used for this negotiation.
. Since the call uses gatekeeper-routed call model, the gatekeeper will usually replace
the H.245 address with it own H.245 address, so that it can inspect H.245 messages as
well. In Figure D, the rewritten H.245 address is denoted with asterisk.
7. The calling endpoint opens a TCP channel to the H.245 address it has received in
the Connect message, and the gatekeeper will establish the second "half" of the
H.245 signalling channel. The endpoints can start exchanging H.245 messages. The
H.245 negotiation has three parts:
• Deciding which endpoint is the "master" and which is the "slave". This has
more of an importance for conferences with multiple participants, rather than
for a two-party call. It is done nonetheless.
• Exchanging information about the capability set of each party. The endpoints
need to know what audio and video codecs the other party supports.
• Deciding about the actual parameters for the audio (and optionally video)
channels - i.e. what codecs will be used and what are the IP addresses and
ports for the RTP streams. This is known as the negotiation of logical
channels.
8. Finally, the two endpoints can start sending the RTP streams and the two
people will hear one another.
The steps 7 and 8 are shown in Figure E.
Let's now have a look at what happens when the call is over:
The two endpoints stop sending the RTP streams. They announce the closing of logical
channels (H.245 Request CloseLogicalChannel).
The H.245 signalling channel is closed (H.245 command
message EndSessionCommand followed by closing of the TCP connection).
The main signalling connection is also closed — the endpoints exchange Q.931/H.225.0
messages ReleaseComplete and the TCP connection is closed.
Each of the two endpoints informs the gatekeeper about the completed call with the H.225.0-
RAS message Disengage Request (DRQ) and the gatekeeper confirms with Disengage
Confirm (DCF).
And now the call is really over!