VoIP allows analog voice signals to be transmitted over the internet by converting voice data into digital packets. At the sender, the analog voice input is converted to digital data using codecs and loaded into IP packets according to protocols. These packets are transmitted over the internet and received, where they are merged back into a binary data stream and converted to analog voice using codecs. Common protocols used include SIP for signaling and RTP within UDP/IP for transmitting voice data packets in real-time. VoIP provides advantages over PSTN like lower costs and increased functionality like conferencing and simultaneous voice and data transmission.

1. Introduction to VoIP
- Tausun Akhtary

2. What is VoIP
● VoIP is short form for Voice over Internet Protocol.
● As the term says VoIP lets analog voice data travel through internet as data packets.

3. How does VoIP work (Brief workflow)
VoIP Adapter
Analog voice input
Source
Digital Data Packets
………...
Internet
Digital Data Packets
Destination
Analog voice output
VoIP Adapter

4. VoIP technical workflow (Sender)
Analog voice input Digital Data Packets
The analog signal is
converted to it’s
digital counterpart
using Voice Codecs
The binary output
data from converter
is loaded into IP
payloads (data
packets) according
to Protocols
Binary
Data
Output
VoIP Adapter/ VoIP Gateway
A/D Converter Packetizer
Internet

5. VoIP technical workflow (Receiver)
Digital Data Packets Analog Voice
Binary
Data
Input
VoIP Adapter/ VoIP Gateway
A/D ConverterPacketizer
Internet
The received data
packets are
merged into binary
data streams
according to the
Protocols
The binary data
stream is
converted to
analog voice data
using Voice
Codecs

6. Explanation (Protocols)
● Protocols - A VoIP protocol determines the frame structures of data packet and how the data
packet is transported across a network.
● Two protocols are necessary for VoIP service:
○ a signaling protocol and
○ a speech transmission protocol.

7. Signaling Protocols
Call signaling is used in Voice over IP (VoIP ) systems to establish connections between endpoints, or
between an endpoint and a gatekeeper. The most commonly used VoIP signaling protocols are as
follows:
❏ SIP (Session Initiation Protocol) – SIP is a standards-based protocol that is used and supported
by the vast majority of VoIP phone systems and services.
❏ SCCP (Cisco Skinny Client Control Protocol) – SCCP is a proprietary protocol used by Cisco's
Call Manager and IP phones.
❏ MGCP – MGCP is an older VoIP protocol. It is no longer widely used and or supported.
❏ H.323 – Similar to MGCP, H.323 is an older VoIP protocol but is no longer widely used and or

8. The raw data is encapsulated into TCP/IP stack following the below structure:
Speech Transmission Protocols
VOIP data packets
RTP
UDP
IP

9. Why UDP
over TCP

10. TCP Header
vs UDP
Header

11. RTP( Real Time Transport Protocol)
● VoIP data packets live in RTP (Real-Time Transport Protocol) packets which are inside UDP-IP
packets.
● Firstly, VoIP doesn't use TCP because it is too heavy for real time applications, so instead a UDP
(datagram) is used.
● Secondly, UDP has no control over the order in which packets arrive at the destination or how
long it takes them to get there (datagram concept). Both of these are very important to overall
voice quality (how well you can understand what the other person is saying) and conversation
quality (how easy it is to carry out a conversation). RTP solves the problem enabling the receiver
to put the packets back into the correct order and not wait too long for packets that have either lost
their way or are taking too long to arrive (we don't need every single voice packet, but we need a
continuous flow of many of them and ordered).

12. RTP Packet Structure
0 1 2 3
0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
|V=2|P|X| CC |M| PT | sequence number |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| timestamp |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
| synchronization source (SSRC) identifier |
+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
| contributing source (CSRC) identifiers |
| .... |
+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+

13. RTP Packet Structure (Contd.)
Where:
● V indicates the version of RTP used
● P indicates the padding, a byte not used at bottom packet to reach the parity packet dimension
● X is the presence of the header extension
● CC field is the number of CSRC identifiers following the fixed header. CSRC field are used, for
example, in conference case.
● M is a marker bit
● PT payload type

14. Explanation( Voice Codecs)
● Voice Codecs - A voice codec is responsible for the compression of the voice stream within a
digital packet. It also determines sound quality and bandwidth required to send the packet.
The most common voice codecs are:
❏ GSM – 13 Kbps
❏ iLBC – 15 Kbps
❏ G.71 1 - 64 Kbps
❏ G.722 - 48/56/64 Kbps
❏ G.726 - 16/24/32/40 Kbps
❏ G.728 - 16 Kbps
❏ G.729 - 8 Kbps

15. Challenges
The user expects a quality of service (QoS) as good as he/she would get on a PSTN dial-up connection.
To achieve this goal, VoIP designers were faced with the following degradations:
❏ Mouth-to-ear delay, Talker echo and Distortion, Silence suppression.
❏ Impact of erred frames (packets) or Lost frames (packets)
❏ Variation of packet arrival time, jitter buffering
❏ Prioritizing VoIP traffic over regular Internet and data services;
❏ Voice coding algorithm standardization;

16. Why Use VoIP
There are two major reasons to use VOIP
● Lower Cost
● Increased functionality

17. VoIP over PSTN
● While using PSTN line, we typically pay for time used to a PSTN line manager company: more
time we stay at phone and more we pay. In opposite with VoIP mechanism we can talk all the time
with every person we want (the needed is that other person is also connected to Internet at the
same time), as far as we want (money independent) .
● In addition we couldn't talk with other than one person at a time. Now we can talk with many
people at the same time with VoIP.
● Moreover, at the same time, we can exchange data with people are you talking with, sending
images, graphs and videos.

18. Thank You :)