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PRESENTED BY
ABDOULIE JOOF
1
6/3/2022
Voice over internet protocol (VoIP) also called IP telephony is a
technology that allows calls to be made using a broadband internet
connection instead of a regular Public Switched Telephone
Network (PSTN) phone line. It allows users to make calls from
their computers, smartphones, or other mobile device, as well as
from VoIP phones and webRTC(real time communication)-enabled
browsers
2
VoIP converts the analog voice signal into a digital signal that can
travels over the internet. When a call is made to a regular telephone
number, the signal is converted to analog just before it reaches the
destination devices.
In VoIP, data packets are used for synchronous voice
communication
3
Since VoIP calls travel over the Internet rather than analog
telephone lines, they experience the same lags and delays as other
data traveling over the Internet when bandwidth is limited or
overloaded.
4
 Skype, Google Voice, Aircall, Zoiper, WhatsApp, Google
Hangouts, Nextiva, Viber, Facebook Messenger etc.
5
There are three types of VOIP communications
 Computer-to-Computer
 Computer-to-Phone
 Phone-to-Phone
6
 REAL-TIME TRANSPORT PROTOCOL (RTP)
 REAL-TIME TRANSPORT CONTROL PROTOCOL (RTCP)
 SESSION INITIATION PROTOCOL (SIP)
 SESSION DESCRIPTION PROTOCOL (SDP)
 H.323 PROTOCOL
7
 RTP was developed by the Audio-Video Transport Working
Group of the Internet Engineering Task Force (IETF)
 This protocol is used to send audio and video across the internet
in a streaming format.
 It has payload type identification, sequence numbering, time
stamping, and delivery tracking features.
8
9
 This protocol is responsible for periodic transmissions of control
packets to all participants in the session.
 It provides feedbacks on the quality of data distribution and
carries a persistent transport level identifier for an RTP source
 It provides feedback on the quality of the data distribution.
10
11
 SIP is used for initiating interactive communication session
between users.
 Provides a direct connection between private or local telephone
system.
 Telephones on the PBX can make or receive calls to/ from PSTN
telephones.
12
13
 SDP is a protocol for describing multimedia sessions for the
purposes of announcement, invitation.
 SDP advertises the details of media e.g. Transport address,
transport protocol, and ports.
14
 Session description – session name and purpose
 Time description - the time the session is active
 Media description - media comprising the session
15
 Is defined by the ITU – T for voice transmission over the internet
16
 Terminal – this represents the end devices. It provides real time two way
communication with another terminal, a gateway or multipoint control units
(MCU). The communication can be voice, data or video.
 Gateway – this establishes the connection between the terminals in the network
and other networks like PSTN or ISDN.
 Gatekeepers – are responsible for translating between telephone numbers and IP
addresses, managing bandwidth and provide mechanisms for terminal
registration and authentication. Gatekeepers also provide services such call
transfer and call forwarding.
 Multipoint control unit (MCU) - takes care of establishing multipoint conferences.
It
consists of a mandatory Multipoint Control, which is used for call signaling and
conference control, and an optional Multipoint Processor, which is used for
switching of the media stream and sometimes real-time transcoding of the
received audio/video streams.
17
 Cheaper calls
 Increased accessibility
 Complete portability
 Higher scalability
 Clearer voice quality
 Conference calls
18
 Reliable Internet Connection Required
 Latency and Jitter
 No location tracking for emergency calls
19
 As PSTN is reaching the end of its life and becoming
increasingly difficult and costly to maintain, the migration to
VoIP is necessary to ensure the continued provision of reliable
landline telephone services globally. VoIP is the future of the
telecoms sector.
20
 In conclusion, VoIP is the technology that allows calls to be
made using a broadband internet connection instead of a regular
Public Switched Telephone Network (PSTN). The most common
VoIP protocols are RTP, RTCP, SIP, SDP and H.323. It provides
cheaper calls, improved accessibility, clearer voice quality and is
therefore the future of the telecommunication sector
21
Thank you !!!
22

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VOIP PROTOCOLS.pptx

  • 2. Voice over internet protocol (VoIP) also called IP telephony is a technology that allows calls to be made using a broadband internet connection instead of a regular Public Switched Telephone Network (PSTN) phone line. It allows users to make calls from their computers, smartphones, or other mobile device, as well as from VoIP phones and webRTC(real time communication)-enabled browsers 2
  • 3. VoIP converts the analog voice signal into a digital signal that can travels over the internet. When a call is made to a regular telephone number, the signal is converted to analog just before it reaches the destination devices. In VoIP, data packets are used for synchronous voice communication 3
  • 4. Since VoIP calls travel over the Internet rather than analog telephone lines, they experience the same lags and delays as other data traveling over the Internet when bandwidth is limited or overloaded. 4
  • 5.  Skype, Google Voice, Aircall, Zoiper, WhatsApp, Google Hangouts, Nextiva, Viber, Facebook Messenger etc. 5
  • 6. There are three types of VOIP communications  Computer-to-Computer  Computer-to-Phone  Phone-to-Phone 6
  • 7.  REAL-TIME TRANSPORT PROTOCOL (RTP)  REAL-TIME TRANSPORT CONTROL PROTOCOL (RTCP)  SESSION INITIATION PROTOCOL (SIP)  SESSION DESCRIPTION PROTOCOL (SDP)  H.323 PROTOCOL 7
  • 8.  RTP was developed by the Audio-Video Transport Working Group of the Internet Engineering Task Force (IETF)  This protocol is used to send audio and video across the internet in a streaming format.  It has payload type identification, sequence numbering, time stamping, and delivery tracking features. 8
  • 9. 9
  • 10.  This protocol is responsible for periodic transmissions of control packets to all participants in the session.  It provides feedbacks on the quality of data distribution and carries a persistent transport level identifier for an RTP source  It provides feedback on the quality of the data distribution. 10
  • 11. 11
  • 12.  SIP is used for initiating interactive communication session between users.  Provides a direct connection between private or local telephone system.  Telephones on the PBX can make or receive calls to/ from PSTN telephones. 12
  • 13. 13
  • 14.  SDP is a protocol for describing multimedia sessions for the purposes of announcement, invitation.  SDP advertises the details of media e.g. Transport address, transport protocol, and ports. 14
  • 15.  Session description – session name and purpose  Time description - the time the session is active  Media description - media comprising the session 15
  • 16.  Is defined by the ITU – T for voice transmission over the internet 16
  • 17.  Terminal – this represents the end devices. It provides real time two way communication with another terminal, a gateway or multipoint control units (MCU). The communication can be voice, data or video.  Gateway – this establishes the connection between the terminals in the network and other networks like PSTN or ISDN.  Gatekeepers – are responsible for translating between telephone numbers and IP addresses, managing bandwidth and provide mechanisms for terminal registration and authentication. Gatekeepers also provide services such call transfer and call forwarding.  Multipoint control unit (MCU) - takes care of establishing multipoint conferences. It consists of a mandatory Multipoint Control, which is used for call signaling and conference control, and an optional Multipoint Processor, which is used for switching of the media stream and sometimes real-time transcoding of the received audio/video streams. 17
  • 18.  Cheaper calls  Increased accessibility  Complete portability  Higher scalability  Clearer voice quality  Conference calls 18
  • 19.  Reliable Internet Connection Required  Latency and Jitter  No location tracking for emergency calls 19
  • 20.  As PSTN is reaching the end of its life and becoming increasingly difficult and costly to maintain, the migration to VoIP is necessary to ensure the continued provision of reliable landline telephone services globally. VoIP is the future of the telecoms sector. 20
  • 21.  In conclusion, VoIP is the technology that allows calls to be made using a broadband internet connection instead of a regular Public Switched Telephone Network (PSTN). The most common VoIP protocols are RTP, RTCP, SIP, SDP and H.323. It provides cheaper calls, improved accessibility, clearer voice quality and is therefore the future of the telecommunication sector 21