Lec40!41!42!43!44_45 Video Conferencing

8/12/2019 Lec40!41!42!43!44_45 Video Conferencing

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INTERACTIVEMULTIMEDIA SYSTEMS

MULTIMEDIA CONFERENCING


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Requirements & BenefitsEach user has a video camera, microphone, andspeakers mounted on his or her computer. As the

participants speak to one another, they hear eachother's voices and see a video image of the other

participant(s). Videoconferencing allows you tohave face-to-face conversations with other peopleon a network, whether they are around the corneror 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


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Application Areas Meetings Education Telemedicine Courts and Judicial System Telecommuting Security

Journalism Television and Media


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THE KEY PIECES All videoconferencing systems have a few key pieces

that make up the system. The components requiredfor 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 acombination of both.


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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 identifiesits contents. The protocol used is usuallydetermined by the need to have reliable orunreliable communications.


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TCP is a reliable protocol designed for transmittingalphanumeric data; it can stop and correct itselfwhen data is lost. This protocol is used to guaranteesequenced, 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 datais lost in preference to maintaining the flow.

TRANSPORT PROTOCOLS


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Real-Time Protocol (RTP) was developed tohandle streaming audio and video and uses IPMulticast.

RTP is a derivative of UDP in which a time-stamp and sequence number is added to the

packet header. This extra information allows thereceiving client to reorder out of sequence

packets, discard duplicates and synchronize audioand video after an initial buffering period.

Real-Time Control Protocol (RTCP) is used to

control RTP .

TRANSPORT PROTOCOLS


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AVAILABLE O PTIONS / MEDIUMS ISDN LAN WAN Internet ADSL (Asynchronous Digital Subscriber Lines) VPN (Virtual Private Networks)

are the popular transport media used in desktopvideo conferencing.

They all have strengths and weaknesses thatshould be considered carefully before decidingupon which one to use.


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The worldwide availability of the Internet hasvirtually stopped the use of POTS (Plain OldTelephone Service) as a direct means ofconnecting video conferencing systems.

However, the forthcoming media-enabled 3Gmobile phone has caused the creation of aderivative of the H.324 POTS standard in the formof 3G-324M as well as next generation Gatewaysto transcode the new protocols.

AVAILABLE OPTIONS


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ISDN In the past, most conferences would have been

between just two participants as ISDN isessentially a point-to-point connection.

However, multipoint technology now makes it possible for groups of people to participate in aconference and share information.

To hold a multipoint conference over ISDN, participants use a Multipoint Control Unit(MCU ), that connects and manages all the ISDNlines.


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H.320 is the ITU standard for ISDN conferencingand 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


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LAN or Intranet and WAN

Unlike ISDN networks, LANs and WANs useTCP/IP protocol and the H.323 standard defineshow to assemble the audio, video, data and

control ( AVDC ) information into an IP packet.

H.323 describes point-to-point and multipointinteroperability of audio and/or visual terminal

equipment connected via a IP basednonguaranteed quality of service network.


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In order to correctly identify a user, the H.323endpoints are usually registered with a Gatekeeperand 'called' into a conference by their H.323 alias.

The Gatekeeper translates the alias into thecorresponding IP address. Another method ofidentifying H.323 users is for them to registertheir presence using Light Directory AccessProtocol (LDAP) with a Directory Service such as

Microsoft's Site Server ILS or Windows 2003Active Directory.



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To hold a multipoint conference over IP, H.323 systemsrequire some form of Multipoint Conference Server(MCS ). This is also referred to as an H.323 MultipointControl Unit ( H.323 MCU).

For small scale multipoint conferences, there are nowendpoints with an embedded H.323 multipointcapability that support up to 4 endpoints in a singleconference.



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H.323 is the ITU standard for LANconferencing and includes:

Audio: G.711, G.722, G.722.1, G.723.1, G.728, G.729Video: H.264, H.263, H.261 Data: H.239, T.120 Control: H.225, H.245



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3G-324M is an extension by the 3rd GenerationPartner Project (3GPP) and 3rd GenerationPartner Project2 (3GPP2) to the ITU H.324Mstandard for 3G mobile phone conferencing andincludes:

Audio: G.722.2 (AMR-WB), G.723.1 Video: MPEG-4 , but not H.264Control: H.223 A/B, H.245


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Video Standards

H.261 - video codec for audiovisual services at p x 64Kbps. P canchange in the range of 1 to 30. Uses RLE encoding, DCT(Discrete Cosine Transform) and motion estimation that requiresrelative low bandwidth. Only non-interlaced video; used inH.320.

H.263 - video codec for narrow telecommunications channels at< 64 Kbps. it's a coding method developed for H.324 that usesH.261 technology with additional improvements. Based on sameDCT and motion compensation technique used in H.261. Notableelements of the standard are image size. QCIF is QuarterCommon Intermediate Format and represents a 176x144 pixelimage. This is the minimum size that must be supported to beH.320 compliant.


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Audio Standards G.711 Does direct sample by sample non-uniform

quantization i.e. Pulse Code Modulation of voicefrequencies (PCM). 3.1 kHz analogue audio is encodedinto a 48, 56 or 64 kbps stream. Used when no otherstandard is equally supported. Default coder for ISDNaudio telephony.

G.722 - 7 kHz audio encoded into a 48, 56 or 64 kbpsstream; used in H.320. Provides high quality, but takes

bandwidth. Divides signal in two passes ( high pass andlow pass) which are then encoded with differentmodalities.

G.722.2 - Coding of speech at around 16 kbps usingAdaptive Multi-Rate Wideband, AMR-WB . Fivemandatory modes, 6.60, 8.85, 12.65, 15.85 and 23.85kbps.


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G.723.1 - 3.4 kHz dual rate speech codec fortelecommunications at 5.3 kbps & 6.4 kbps.

G.728 - 3.4 kHz Low Delay Code Excited LinearPrediction (LD-CELP) were 3.4 kHz analogue audio isencoded into a 16 kbps stream. This standard providesgood quality results at low bit rates. Low delay buthigh complexity. Suggested speech coder for low bit

rate (64-128 kbps) ISDN telephony.

T.120 - defines protocols and services for multimediaconferencing


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Videoconference etiquette Prepare and distribute an agenda in advance.

Reading facial expressions and body language are thenext most important parts of a conversation, so set thecamera view in proper way.

Avoid wearing Bright colors. Avoid bold, complex or busy patterns like small checks or narrow stripes inclothing, scarves.

Listen for environmental noise such as fans, openwindows, pens clicking, and papers shuffling thatmight 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

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