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INTERACTIVE
MULTIMEDIA SYSTEMS
MULTIMEDIA CONFERENCING
Introduction
• Videoconferencing is exchange of digital video
images and sound among two or more distant
parties.
• Transferred images can include video streams,
immovable images of objects, and data from
graphics, files or applications. It allows participants
to hear, to see and to collaborate in the real time
mode with all their interlocutors.
Requirements & Benefits
Each user has a video camera, microphone, and
speakers mounted on his or her computer. As the
participants speak to one another, they hear each
other's voices and see a video image of the other
participant(s). Videoconferencing allows you to
have face-to-face conversations with other people
on a network, whether they are around the corner
or on the other side of the world.
Benefits
• Visual interaction enhances communication
• Increases connections with the world
• It can save time and other valuable resources
Application Areas
• Meetings
• Education
• Telemedicine
• Courts and Judicial System
• Telecommuting
• Security
• Journalism
• Television and Media
THE KEY PIECES
• All videoconferencing systems have a few key pieces
that make up the system. The components required
for a videoconference system include:
• Video Input: video camera or webcam
• Video Output: computer monitor or television
• Audio Input: microphones
• Audio Output: loudspeakers associated with the
display device or telephone or headphones
• Data Transfer: analog or digital telephone network,
LAN or Internet.
• Codec: In simple terms, a codec is any technology
for compressing and decompressing data. Codecs can
be implemented in software, hardware, or a
combination of both.
TRANSPORT PROTOCOLS
• There are several standards based transport
protocols used with conferencing, TCP, UDP &
RTP.
• Generally, each configures the data into packets,
with each packet having a 'header' that identifies
its contents. The protocol used is usually
determined by the need to have reliable or
unreliable communications.
• TCP is a reliable protocol designed for transmitting
alphanumeric data; it can stop and correct itself
when data is lost. This protocol is used to guarantee
sequenced, error-free transmission, but its very
nature can cause delays and reduced throughput.
This can be irritating, especially with audio.
• User Datagram Protocol (UDP) within the IP stack
is by contrast, an unreliable protocol in which data
is lost in preference to maintaining the flow.
TRANSPORT PROTOCOLS
• Real-Time Protocol (RTP) was developed to
handle streaming audio and video and uses IP
Multicast.
• RTP is a derivative of UDP in which a time-
stamp and sequence number is added to the
packet header. This extra information allows the
receiving client to reorder out of sequence
packets, discard duplicates and synchronize audio
and video after an initial buffering period.
• Real-Time Control Protocol (RTCP) is used to
control RTP.
TRANSPORT PROTOCOLS
AVAILABLE OPTIONS / MEDIUMS
• ISDN
• LAN
• WAN
• Internet
• ADSL (Asynchronous Digital Subscriber Lines)
• VPN (Virtual Private Networks)
are the popular transport media used in desktop
video conferencing.
• They all have strengths and weaknesses that
should be considered carefully before deciding
upon which one to use.
• The worldwide availability of the Internet has
virtually stopped the use of POTS (Plain Old
Telephone Service) as a direct means of
connecting video conferencing systems.
• However, the forthcoming media-enabled 3G
mobile phone has caused the creation of a
derivative of the H.324 POTS standard in the
form of 3G-324M as well as next generation
Gateways to transcode the new protocols.
AVAILABLE OPTIONS
TYPES OF
VIDEO CONFERENCING
Traditional Internet
Use in special room; Use anywhere;
Uses ISDN telephone lines Uses Internet
High installation cost Low installation cost
High usage cost No usage cost
Usage at plateau Usage growing rapidly
Professional operator Do-it-yourself
Centralized control Decentralized control
H.320 standard H.323 standard
Video Conferencing
Standards
ISDN
• In the past, most conferences would have been
between just two participants as ISDN is
essentially a point-to-point connection.
• However, multipoint technology now makes it
possible for groups of people to participate in a
conference and share information.
• To hold a multipoint conference over ISDN,
participants use a Multipoint Control Unit
(MCU), that connects and manages all the ISDN
lines.
• H.320 is the ITU standard for ISDN conferencing
and includes:
Audio: G.711, G.722, G.722.1, G.728
Video: H.264, H.263, H.261
Data: T.120
Control: H.221, H.231, H.242, H.243
ISDN
LAN or Intranet and WAN
• Unlike ISDN networks, LANs and WANs use
TCP/IP protocol and the H.323 standard defines
how to assemble the audio, video, data and
control (AVDC) information into an IP packet.
• H.323 describes point-to-point and multipoint
interoperability of audio and/or visual terminal
equipment connected via a IP based
nonguaranteed quality of service network.
• In order to correctly identify a user, the H.323
endpoints are usually registered with a Gatekeeper
and 'called' into a conference by their H.323 alias.
• The Gatekeeper translates the alias into the
corresponding IP address. Another method of
identifying H.323 users is for them to register
their presence using Light Directory Access
Protocol (LDAP) with a Directory Service such as
Microsoft's Site Server ILS or Windows 2003
Active Directory.
LAN or Intranet and WAN
• To hold a multipoint conference over IP, H.323
systems require some form of Multipoint Conference
Server (MCS). This is also referred to as an H.323
Multipoint Control Unit (H.323 MCU).
• For small scale multipoint conferences, there are now
endpoints with an embedded H.323 multipoint
capability that support up to 4 endpoints in a single
conference.
LAN or Intranet and WAN
• H.323 is the ITU standard for LAN
conferencing and includes:
Audio: G.711, G.722, G.722.1, G.723.1, G.728, G.729
Video: H.264, H.263, H.261
Data: H.239, T.120
Control: H.225, H.245
LAN or Intranet and WAN
GateKeeper
H.323 Clients
INTERNET
H.323 Videoconferencing
Web
Browser
Real Video
H.323
H.320
Gateway
ISDN
POTS line
Telephone
MCU
Gateway
Plain Old Telephone Service (POTS)
• The standard telephone system is the most readily
available form of transport media for home users.
• With V.92 modems giving transmission speeds of up
to 56kbps, there is just about sufficient bandwidth
available to support audio, video and data sharing
with this media, especially when used in conjunction
with the latest CPU's, compression techniques and
technologies like DirectDraw.
• However, the use of standard telephone lines for
POTS based H.324 conferencing has given way to the
ever increasing popularity of the Internet.
Cellular Networks
• The cellular phone network is a readily available form
of wireless multimedia delivery and with the
forthcoming media-enabled 3G mobile phone or
Personal Digital Assistants, PDAs, that support the
CDMA2000 or WCDMA Air Interface, there is
sufficient bandwidth to enable IP-based multipoint
audio and video conferencing to existing desktop video
conferencing systems when used in-conjunction with
next generation Gateways and MCU's that also support
these new protocols.
• 3G-324M is an extension by the 3rd Generation
Partner Project (3GPP) and 3rd Generation
Partner Project2 (3GPP2) to the ITU H.324M
standard for 3G mobile phone conferencing and
includes:
Audio: G.722.2 (AMR-WB), G.723.1
Video: MPEG-4, but not H.264
Control: H.223 A/B, H.245
Video Standards
• H.261 - video codec for audiovisual services at p x 64Kbps. P
can change in the range of 1 to 30. Uses RLE encoding, DCT
(Discrete Cosine Transform) and motion estimation that requires
relative low bandwidth. Only non-interlaced video; used in
H.320.
• H.263 - video codec for narrow telecommunications channels at
< 64 Kbps. it's a coding method developed for H.324 that uses
H.261 technology with additional improvements. Based on same
DCT and motion compensation technique used in H.261. Notable
elements of the standard are image size. QCIF is Quarter
Common Intermediate Format and represents a 176x144 pixel
image. This is the minimum size that must be supported to be
H.320 compliant.
Audio Standards
• G.711 – Does direct sample by sample non-uniform
quantization i.e. Pulse Code Modulation of voice
frequencies (PCM). 3.1 kHz analogue audio is encoded
into a 48, 56 or 64 kbps stream. Used when no other
standard is equally supported. Default coder for ISDN
audio telephony.
• G.722 - 7 kHz audio encoded into a 48, 56 or 64 kbps
stream; used in H.320. Provides high quality, but takes
bandwidth. Divides signal in two passes ( high pass and
low pass) which are then encoded with different
modalities.
• G.722.2 - Coding of speech at around 16 kbps using
Adaptive Multi-Rate Wideband, AMR-WB. Five
mandatory modes, 6.60, 8.85, 12.65, 15.85 and 23.85
kbps.
• G.723.1 - 3.4 kHz dual rate speech codec for
telecommunications at 5.3 kbps & 6.4 kbps.
• G.728 - 3.4 kHz Low Delay Code Excited Linear
Prediction (LD-CELP) were 3.4 kHz analogue audio is
encoded into a 16 kbps stream. This standard provides
good quality results at low bit rates. Low delay but
high complexity. Suggested speech coder for low bit
rate (64-128 kbps) ISDN telephony.
• T.120 - defines protocols and services for multimedia
conferencing
Videoconference etiquette
• Prepare and distribute an agenda in advance.
• Reading facial expressions and body language are the
next most important parts of a conversation, so set the
camera view in proper way.
• Avoid wearing Bright colors. Avoid bold, complex or
busy patterns like small checks or narrow stripes in
clothing, scarves.
• Listen for environmental noise such as fans, open
windows, pens clicking, and papers shuffling that
might disturb your audio quality.
• Keep body movements to a minimum. Avoid
distracting movements like swaying, or rocking.

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Lec40 41 42_43_44_45 video conferencing

  • 2. Introduction • Videoconferencing is exchange of digital video images and sound among two or more distant parties. • Transferred images can include video streams, immovable images of objects, and data from graphics, files or applications. It allows participants to hear, to see and to collaborate in the real time mode with all their interlocutors.
  • 3. Requirements & Benefits Each user has a video camera, microphone, and speakers mounted on his or her computer. As the participants speak to one another, they hear each other's voices and see a video image of the other participant(s). Videoconferencing allows you to have face-to-face conversations with other people on a network, whether they are around the corner or on the other side of the world. Benefits • Visual interaction enhances communication • Increases connections with the world • It can save time and other valuable resources
  • 4. Application Areas • Meetings • Education • Telemedicine • Courts and Judicial System • Telecommuting • Security • Journalism • Television and Media
  • 5. THE KEY PIECES • All videoconferencing systems have a few key pieces that make up the system. The components required for a videoconference system include: • Video Input: video camera or webcam • Video Output: computer monitor or television • Audio Input: microphones • Audio Output: loudspeakers associated with the display device or telephone or headphones • Data Transfer: analog or digital telephone network, LAN or Internet. • Codec: In simple terms, a codec is any technology for compressing and decompressing data. Codecs can be implemented in software, hardware, or a combination of both.
  • 6. TRANSPORT PROTOCOLS • There are several standards based transport protocols used with conferencing, TCP, UDP & RTP. • Generally, each configures the data into packets, with each packet having a 'header' that identifies its contents. The protocol used is usually determined by the need to have reliable or unreliable communications.
  • 7. • TCP is a reliable protocol designed for transmitting alphanumeric data; it can stop and correct itself when data is lost. This protocol is used to guarantee sequenced, error-free transmission, but its very nature can cause delays and reduced throughput. This can be irritating, especially with audio. • User Datagram Protocol (UDP) within the IP stack is by contrast, an unreliable protocol in which data is lost in preference to maintaining the flow. TRANSPORT PROTOCOLS
  • 8. • Real-Time Protocol (RTP) was developed to handle streaming audio and video and uses IP Multicast. • RTP is a derivative of UDP in which a time- stamp and sequence number is added to the packet header. This extra information allows the receiving client to reorder out of sequence packets, discard duplicates and synchronize audio and video after an initial buffering period. • Real-Time Control Protocol (RTCP) is used to control RTP. TRANSPORT PROTOCOLS
  • 9. AVAILABLE OPTIONS / MEDIUMS • ISDN • LAN • WAN • Internet • ADSL (Asynchronous Digital Subscriber Lines) • VPN (Virtual Private Networks) are the popular transport media used in desktop video conferencing. • They all have strengths and weaknesses that should be considered carefully before deciding upon which one to use.
  • 10. • The worldwide availability of the Internet has virtually stopped the use of POTS (Plain Old Telephone Service) as a direct means of connecting video conferencing systems. • However, the forthcoming media-enabled 3G mobile phone has caused the creation of a derivative of the H.324 POTS standard in the form of 3G-324M as well as next generation Gateways to transcode the new protocols. AVAILABLE OPTIONS
  • 11. TYPES OF VIDEO CONFERENCING Traditional Internet Use in special room; Use anywhere; Uses ISDN telephone lines Uses Internet High installation cost Low installation cost High usage cost No usage cost Usage at plateau Usage growing rapidly Professional operator Do-it-yourself Centralized control Decentralized control H.320 standard H.323 standard
  • 13. ISDN • In the past, most conferences would have been between just two participants as ISDN is essentially a point-to-point connection. • However, multipoint technology now makes it possible for groups of people to participate in a conference and share information. • To hold a multipoint conference over ISDN, participants use a Multipoint Control Unit (MCU), that connects and manages all the ISDN lines.
  • 14. • H.320 is the ITU standard for ISDN conferencing and includes: Audio: G.711, G.722, G.722.1, G.728 Video: H.264, H.263, H.261 Data: T.120 Control: H.221, H.231, H.242, H.243 ISDN
  • 15. LAN or Intranet and WAN • Unlike ISDN networks, LANs and WANs use TCP/IP protocol and the H.323 standard defines how to assemble the audio, video, data and control (AVDC) information into an IP packet. • H.323 describes point-to-point and multipoint interoperability of audio and/or visual terminal equipment connected via a IP based nonguaranteed quality of service network.
  • 16. • In order to correctly identify a user, the H.323 endpoints are usually registered with a Gatekeeper and 'called' into a conference by their H.323 alias. • The Gatekeeper translates the alias into the corresponding IP address. Another method of identifying H.323 users is for them to register their presence using Light Directory Access Protocol (LDAP) with a Directory Service such as Microsoft's Site Server ILS or Windows 2003 Active Directory. LAN or Intranet and WAN
  • 17. • To hold a multipoint conference over IP, H.323 systems require some form of Multipoint Conference Server (MCS). This is also referred to as an H.323 Multipoint Control Unit (H.323 MCU). • For small scale multipoint conferences, there are now endpoints with an embedded H.323 multipoint capability that support up to 4 endpoints in a single conference. LAN or Intranet and WAN
  • 18. • H.323 is the ITU standard for LAN conferencing and includes: Audio: G.711, G.722, G.722.1, G.723.1, G.728, G.729 Video: H.264, H.263, H.261 Data: H.239, T.120 Control: H.225, H.245 LAN or Intranet and WAN
  • 19. GateKeeper H.323 Clients INTERNET H.323 Videoconferencing Web Browser Real Video H.323 H.320 Gateway ISDN POTS line Telephone MCU Gateway
  • 20. Plain Old Telephone Service (POTS) • The standard telephone system is the most readily available form of transport media for home users. • With V.92 modems giving transmission speeds of up to 56kbps, there is just about sufficient bandwidth available to support audio, video and data sharing with this media, especially when used in conjunction with the latest CPU's, compression techniques and technologies like DirectDraw. • However, the use of standard telephone lines for POTS based H.324 conferencing has given way to the ever increasing popularity of the Internet.
  • 21. Cellular Networks • The cellular phone network is a readily available form of wireless multimedia delivery and with the forthcoming media-enabled 3G mobile phone or Personal Digital Assistants, PDAs, that support the CDMA2000 or WCDMA Air Interface, there is sufficient bandwidth to enable IP-based multipoint audio and video conferencing to existing desktop video conferencing systems when used in-conjunction with next generation Gateways and MCU's that also support these new protocols.
  • 22. • 3G-324M is an extension by the 3rd Generation Partner Project (3GPP) and 3rd Generation Partner Project2 (3GPP2) to the ITU H.324M standard for 3G mobile phone conferencing and includes: Audio: G.722.2 (AMR-WB), G.723.1 Video: MPEG-4, but not H.264 Control: H.223 A/B, H.245
  • 23. Video Standards • H.261 - video codec for audiovisual services at p x 64Kbps. P can change in the range of 1 to 30. Uses RLE encoding, DCT (Discrete Cosine Transform) and motion estimation that requires relative low bandwidth. Only non-interlaced video; used in H.320. • H.263 - video codec for narrow telecommunications channels at < 64 Kbps. it's a coding method developed for H.324 that uses H.261 technology with additional improvements. Based on same DCT and motion compensation technique used in H.261. Notable elements of the standard are image size. QCIF is Quarter Common Intermediate Format and represents a 176x144 pixel image. This is the minimum size that must be supported to be H.320 compliant.
  • 24. Audio Standards • G.711 – Does direct sample by sample non-uniform quantization i.e. Pulse Code Modulation of voice frequencies (PCM). 3.1 kHz analogue audio is encoded into a 48, 56 or 64 kbps stream. Used when no other standard is equally supported. Default coder for ISDN audio telephony. • G.722 - 7 kHz audio encoded into a 48, 56 or 64 kbps stream; used in H.320. Provides high quality, but takes bandwidth. Divides signal in two passes ( high pass and low pass) which are then encoded with different modalities. • G.722.2 - Coding of speech at around 16 kbps using Adaptive Multi-Rate Wideband, AMR-WB. Five mandatory modes, 6.60, 8.85, 12.65, 15.85 and 23.85 kbps.
  • 25. • G.723.1 - 3.4 kHz dual rate speech codec for telecommunications at 5.3 kbps & 6.4 kbps. • G.728 - 3.4 kHz Low Delay Code Excited Linear Prediction (LD-CELP) were 3.4 kHz analogue audio is encoded into a 16 kbps stream. This standard provides good quality results at low bit rates. Low delay but high complexity. Suggested speech coder for low bit rate (64-128 kbps) ISDN telephony. • T.120 - defines protocols and services for multimedia conferencing
  • 26. Videoconference etiquette • Prepare and distribute an agenda in advance. • Reading facial expressions and body language are the next most important parts of a conversation, so set the camera view in proper way. • Avoid wearing Bright colors. Avoid bold, complex or busy patterns like small checks or narrow stripes in clothing, scarves. • Listen for environmental noise such as fans, open windows, pens clicking, and papers shuffling that might disturb your audio quality. • Keep body movements to a minimum. Avoid distracting movements like swaying, or rocking.

Notas del editor

  1. Basic Equipment Components Codec: In simple terms, a codec is any technology for compressing and decompressing data. Codecs can be implemented in software, hardware, or a combination of both.The codec takes the analog video signal and codes (digitizes and compresses) it. The codec also has to decode (decompress and un-digitize) the received transmission. You can imagine that this kind of processing can take its toll on the video and sound quality. The most obvious consequence of a slow codec or low-bandwidth connection is a &amp;quot;jerky&amp;quot; picture and an audio time delay. We discuss how to deal with compressed video features in our Working With Compressed Video section. For more technical information about codecs, try this site: IP Video Endpoint (Codec) from the Indiana Higher Education Telecommunication System (IHETS). Monitor: Desktop systems display video in a small window on the computer monitor much like the movies and other media that we are used to playing on our computers. Portable and fixed systems have one or two large video monitors and usually display the local audience as well as the remote audience. In conference-type settings, large movie or presentation screens can be used and the video is projected onto the screen. Strings of monitors are also used in conference-type settings. Camera: The camera can be anything from a tiny desktop camera that sits on top of a computer monitor (desktop system) to a high-quality model with remote control pan and zoom features (room system). High-end systems often come with a variety of input sockets allowing for connections to other peripherals such as a document camera, a second video input connection, and other auxilary equipment. When you expect to do a question and answer session, using two cameras is especially desirable. That way, the facilitator or main speaker has one dedicated camera and another can be used to pan the audience or pull in tight on a person asking a question. The second camera doesn&amp;apos;t even need to be fully functional -- all it has to do is pass the input to the system! Audio: Most high-quality systems come with a microphone designed for use with a small group of people. In many cases, an additional microphone can be connected as well, making your setup more versitile for larger groups or to add mobility. Most systems offer sophisticated dianostics and processing as a built-in feature to cancel out background noise and echo. Control: Controls allow users to place calls, adjust volume, and sometimes even pan and zoom the camera. Desktop systems display controls and tools on the computer monitor window. Room systems come with remote control or console devices. Source: http://www.kn.pacbell.com/wired/vidconf/equipment.html