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Voice Over IP (VoIP)
&
STREAMING
Realised by:
- Ragheb Gmira
- Faten barkati
4C_Info2 University year 2015/2016
Outline
 VOIP
 Definition
 How does it work?
 VoIP architecture
 Advantages & limits
 Protocols
 STREAMING
 Definition
 How does it work?
 Protocols
 Avantages
 Desavantages
 Conclusion
VOIP
 Definition
 Voice over Internet Protocol (VoIP), is a technology that allows to make voice calls using
a broadband Internet connection instead of a regular (or analog) phone line (PSTN).
 The use of IP networks, namely the LAN and WAN, to carry voice,
 Voice conversations are turned into digitized data and packetized for transmission
across a network.
How does it work?
 VoIP services convert your voice into a digital signal that travels over the Internet. If
you are calling a regular phone number, the signal is converted to a regular telephone
signal before it reaches the destination. VoIP can allow to make a call directly from a
computer, a special VoIP phone, or a traditional phone connected to a special adapter.
 Analog Telephone Adaptor : sometimes called a ‘gateway’, is a device for connecting traditional
analog telephones, fax machines, and similar customer-premises devices to a digital telephone
system or a voice over IP telephony network.
 select whether outgoing calls use VoIP or the standard telephone connection
 switch between VoIP and standard phone connection for incoming calls.
 VoIP phone : A dedicated VOIP handset looks just like a standard phone, but it is connected
directly to router or PC, typically through an Ethernet or USB port. The handset already includes
all software necessary to send and receive VoIP calls
 Computer to computer using a Software VoIP Phone : As the term suggests, this involves using
software on PC and an internet connection to contact someone else with compatible software at
their computer. Apart from the software, the computer needs a microphone, speakers, a suitable
sound card and a fast internet connection.
VoIP phone calls can be made in one of three ways.
 VoIP architecture
PSTN : Public switched telephone network PABX : Private Automatic Branch eXchange
 Advantages & limits
 Advantages:
 You don’t need to pay the additional cost for a land-line.
 Many services allow you to talk for as long as you want.
 It is easily portable and can be used in any destination.
 A triple play network: it includes voice, video and data.
 Many conversations at the same time.
 Exchange data with people are you talking with, sending images, graphs and
videos.
 Limits:
 The needed is that other person is also connected to Internet at the same time.
 Voice data communication must be a real time stream
 Protocols
 The protocol H.323 : (Packet-based multimedia communications systems)
a multimedia standard that provides a foundation to transport voice, video and data
with a concern for real-time data.
H.323 defines several network elements:
o Terminals (LAN Endpoints) :IP phone and PC
o Gateways (Optional but really useful) : Device that connects H.323 voice network
to non-H.323 voice network (SIP or PSTN)
o Gatekeepers (Also optional)
• Provides address translation (phone number to IP address)
• Admission control for H.323 terminals and gateways
• Manage bandwidth allocation
• Other optional services (LAN traffic management)
o MCUs (multipoint control units) : is a device that is used for multiparty
conferencing
 The protocol SIP :
o It is an application layer control protocol for creating, modifying and terminating
sessions with one or more participants. The architecture of SIP is similar to that of
HTTP (client-server protocol). Requests are generated by the client and sent to the
serverSIP ,
The SIP System consists ot two components : User Agents and Network Servers
SIP defines a lot of messages :
• INVITE: for inviting a user to a call
• BYE: for terminating a connection between the two end points
• ACK: for reliable exchange of invitation messages
• OPTIONS: for getting information about the capabilities of a call
• REGISTER: gives information about the location of a user to the SIP registration
server.
• CANCEL: for terminating the search for a user
 The protocol RTP : Real-time Transport Protocol
o Reconstructing the time base flows.
o Sequencing packages.
o Identify the data content for their associate a secure transport.
o Identify the data source.
o Transport of audio and video applications in frame.
 The protocol RTCP: Real-time Control Protocol
o Periodic spontaneous trasmission of control packets to all participant.
o Feed back on the quality of data distribution
o useful to sender (e.g to adapt encoding)
o useful to receivers (e.g to evaluate whether problems are local)
o Evaluate the participants and adjust report rate
The protocol SIP
STREAMING
A technique for transferring data such that it can be processed
as a steady and continuous stream, client does not have to download
the entire file to view it.
Data
Source
Network
The portion client is viewing
The portion in the buffer
How does it work?
 When audio or video is streamed, a small buffer space is created on the user's
computer, and data starts downloading into it. As soon as the buffer is full (usually just
a matter of seconds), the file starts to play. As the file plays, it uses up information in
the buffer, but while it is playing, more data is being downloaded. As long as the data
can be downloaded as fast as it is used up in playback, the file will play smoothly.
 How does it work?
 Types of streaming
 Streaming Stored Audio and Video : The multimedia content has been prerecorded and stored on a server
• User may pause, rewind, forward, etc…
• Constraint: after display start, the playout must be continuous
 Streaming Live Audio and Video :
• Similar to traditional broadcast TV/radio, but
delivery on the Internet
• Non-interactive just view/listen
• Can not pause or rewind
• Often combined with multicast
 Real-Time Interactive Audio and Video :Phone conversation/Video conferencing
• Constraint: delay between initial request and display start must be small
• Constraint: after display start, the playout must be continuous
 Protocols
 RTP : Real-time transfer Protocol
 RTCP : Real-time transport Control Protocol
 RTSP: Real-time Streaming Protocol
 Real Time Streaming Protocol (RTSP)
RTSP is the control protocol for the delivery of multimedia content across IP networks. It is
based typically on TCP for reliable delivery and has a very similar operation and syntax
to HTTP. RTSP is used by the client application to communicate to the server information
such as the media file being requested
Avantages
 It reduces download time
 It provides steady service
 Slower systems can take advantage of streaming technology
 Provide service on demand
Disadvantages
 It is difficult to keep the service steady if the bandwidth is low
 The maintenance of streaming server is relatively expensive
 Packet loss may occur during the transmission
Conclusion
 VOIP & Streaming are being developped day by day.
 We can probably assume that IP will become one day a single standard
 allowing interoperability of networks globalized.
 That is why the integration of voice over IP is just one step towards EoIP:
Everything over IP
Voice Over IP (VoIP) and STREAMING

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Voice Over IP (VoIP) and STREAMING

  • 1. Voice Over IP (VoIP) & STREAMING Realised by: - Ragheb Gmira - Faten barkati 4C_Info2 University year 2015/2016
  • 2. Outline  VOIP  Definition  How does it work?  VoIP architecture  Advantages & limits  Protocols  STREAMING  Definition  How does it work?  Protocols  Avantages  Desavantages  Conclusion
  • 3. VOIP  Definition  Voice over Internet Protocol (VoIP), is a technology that allows to make voice calls using a broadband Internet connection instead of a regular (or analog) phone line (PSTN).  The use of IP networks, namely the LAN and WAN, to carry voice,  Voice conversations are turned into digitized data and packetized for transmission across a network.
  • 4. How does it work?  VoIP services convert your voice into a digital signal that travels over the Internet. If you are calling a regular phone number, the signal is converted to a regular telephone signal before it reaches the destination. VoIP can allow to make a call directly from a computer, a special VoIP phone, or a traditional phone connected to a special adapter.
  • 5.  Analog Telephone Adaptor : sometimes called a ‘gateway’, is a device for connecting traditional analog telephones, fax machines, and similar customer-premises devices to a digital telephone system or a voice over IP telephony network.  select whether outgoing calls use VoIP or the standard telephone connection  switch between VoIP and standard phone connection for incoming calls.  VoIP phone : A dedicated VOIP handset looks just like a standard phone, but it is connected directly to router or PC, typically through an Ethernet or USB port. The handset already includes all software necessary to send and receive VoIP calls  Computer to computer using a Software VoIP Phone : As the term suggests, this involves using software on PC and an internet connection to contact someone else with compatible software at their computer. Apart from the software, the computer needs a microphone, speakers, a suitable sound card and a fast internet connection. VoIP phone calls can be made in one of three ways.
  • 6.  VoIP architecture PSTN : Public switched telephone network PABX : Private Automatic Branch eXchange
  • 7.  Advantages & limits  Advantages:  You don’t need to pay the additional cost for a land-line.  Many services allow you to talk for as long as you want.  It is easily portable and can be used in any destination.  A triple play network: it includes voice, video and data.  Many conversations at the same time.  Exchange data with people are you talking with, sending images, graphs and videos.  Limits:  The needed is that other person is also connected to Internet at the same time.  Voice data communication must be a real time stream
  • 8.  Protocols  The protocol H.323 : (Packet-based multimedia communications systems) a multimedia standard that provides a foundation to transport voice, video and data with a concern for real-time data. H.323 defines several network elements: o Terminals (LAN Endpoints) :IP phone and PC o Gateways (Optional but really useful) : Device that connects H.323 voice network to non-H.323 voice network (SIP or PSTN) o Gatekeepers (Also optional) • Provides address translation (phone number to IP address) • Admission control for H.323 terminals and gateways • Manage bandwidth allocation • Other optional services (LAN traffic management) o MCUs (multipoint control units) : is a device that is used for multiparty conferencing
  • 9.  The protocol SIP : o It is an application layer control protocol for creating, modifying and terminating sessions with one or more participants. The architecture of SIP is similar to that of HTTP (client-server protocol). Requests are generated by the client and sent to the serverSIP , The SIP System consists ot two components : User Agents and Network Servers SIP defines a lot of messages : • INVITE: for inviting a user to a call • BYE: for terminating a connection between the two end points • ACK: for reliable exchange of invitation messages • OPTIONS: for getting information about the capabilities of a call • REGISTER: gives information about the location of a user to the SIP registration server. • CANCEL: for terminating the search for a user
  • 10.  The protocol RTP : Real-time Transport Protocol o Reconstructing the time base flows. o Sequencing packages. o Identify the data content for their associate a secure transport. o Identify the data source. o Transport of audio and video applications in frame.
  • 11.  The protocol RTCP: Real-time Control Protocol o Periodic spontaneous trasmission of control packets to all participant. o Feed back on the quality of data distribution o useful to sender (e.g to adapt encoding) o useful to receivers (e.g to evaluate whether problems are local) o Evaluate the participants and adjust report rate
  • 13. STREAMING A technique for transferring data such that it can be processed as a steady and continuous stream, client does not have to download the entire file to view it. Data Source Network The portion client is viewing The portion in the buffer
  • 14. How does it work?  When audio or video is streamed, a small buffer space is created on the user's computer, and data starts downloading into it. As soon as the buffer is full (usually just a matter of seconds), the file starts to play. As the file plays, it uses up information in the buffer, but while it is playing, more data is being downloaded. As long as the data can be downloaded as fast as it is used up in playback, the file will play smoothly.
  • 15.  How does it work?
  • 16.  Types of streaming  Streaming Stored Audio and Video : The multimedia content has been prerecorded and stored on a server • User may pause, rewind, forward, etc… • Constraint: after display start, the playout must be continuous  Streaming Live Audio and Video : • Similar to traditional broadcast TV/radio, but delivery on the Internet • Non-interactive just view/listen • Can not pause or rewind • Often combined with multicast  Real-Time Interactive Audio and Video :Phone conversation/Video conferencing • Constraint: delay between initial request and display start must be small • Constraint: after display start, the playout must be continuous
  • 17.  Protocols  RTP : Real-time transfer Protocol  RTCP : Real-time transport Control Protocol  RTSP: Real-time Streaming Protocol
  • 18.  Real Time Streaming Protocol (RTSP) RTSP is the control protocol for the delivery of multimedia content across IP networks. It is based typically on TCP for reliable delivery and has a very similar operation and syntax to HTTP. RTSP is used by the client application to communicate to the server information such as the media file being requested
  • 19. Avantages  It reduces download time  It provides steady service  Slower systems can take advantage of streaming technology  Provide service on demand
  • 20. Disadvantages  It is difficult to keep the service steady if the bandwidth is low  The maintenance of streaming server is relatively expensive  Packet loss may occur during the transmission
  • 21. Conclusion  VOIP & Streaming are being developped day by day.  We can probably assume that IP will become one day a single standard  allowing interoperability of networks globalized.  That is why the integration of voice over IP is just one step towards EoIP: Everything over IP