HD Video Conference Equipments - Peoplelink - …...PeopleLink videoconference system Technical...

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Technical Whitepaper

NextGen-MultiLocation

HD TelePresence Grade Solution

“Single Solution which allows you connect 2-1000 locations in

FullHD 1080P from different environments Like Boardrooms,

Meeting Rooms, Auditoriums, Mobile users from Laptops with data

Cards or Android based Mobile Phone or iPhones and also

Connects to normal PSTN, VOIP and SIP Phones or Epabx’s for

Audio calls with Complete Data Collaboration Suite and Recording

and also connects to H.323 based Hardware VC end Points and

MCU’s“

PeopleLink Video Conference Technical Whitepaper Index

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Index

Chapter 1 Product Introduction ........................................................................................ 4 Chapter 2 System Functionalities ..................................................................................... 5

2.1 System Highlights ........................................................................................................ 5 2.2 Audio Video Highlights ................................................................................................ 7 2.3 Data Collaboration Features ....................................................................................... 8 2.4 Conference Management .......................................................................................... 10 2.5 Connection Capability ................................................................................................ 11

Chapter 3 Conference Server System Structure ........................................................ 13 3.1 JabberD ....................................................................................................................... 14 3.2 MCU ............................................................................................................................. 14 3.3 Multimedia Server ...................................................................................................... 14 3.4 File Server ................................................................................................................... 14 3.5 Whiteboard Server ..................................................................................................... 14 3.6 Application Sharing (AppSharing) Server ............................................................... 15 3.7 Conference Management Server ............................................................................. 15 3.8 C2S Module (Optional) .............................................................................................. 15

Chapter 4 Communication Standards ............................................................................ 17 4.1 Communication Protocol: IETF XMPP ..................................................................... 17

4.1.1 Distributed Network ...................................................................................... 17 4.1.2 C/S Structure ................................................................................................. 18 4.1.3 Server Modularity .......................................................................................... 18 4.1.4 XML Format .................................................................................................... 19

4.2 Audio/Video Transmission Protocol: RTP ................................................................ 19 4.3 Audio/Video Transmission Protocol: HTTP ............................................................. 19

Chapter 5 Main Communication Standards ................................................................. 21 5.1 Video Codec: H.264VBR/CBR ................................... Error! Bookmark not defined.

5.1.2 H.263, H264 Series of Standards ............................................................... 21 5.1.3 Reason for using H.264VBR/CBR ................................................................ 21 5.1.4 Future of Video Compression ...................................................................... 22

5.2 Audio Codec: G.723.1 ............................................................................................... 22 5.2.1 G.723.1 Audio Quality .................................................................................. 22 5.2.2 Why use G.723.1 for audio compression ................................................... 23

5.3 Packet Lost Reduction Algorithm ............................................................................. 23 5.4 Bandwidth Adaptability ............................................................................................. 24 5.5 Video Frame Rate ...................................................................................................... 24 5.6 Desktop/Application Sharing Bandwidth Adaptability ........................................... 25 5.7 High Quality Broadband Video ................................................................................. 25 5.8 Hardware Selection .................................................................................................... 26

5.8.1 Desktop/Laptop Setup .................................................................................. 26 5.8.2 Conference Room Setup .............................................................................. 26

5.9 Communication with Videoconferencing Hardware .............................................. 27

PeopleLink videoconference system Technical Whitepaper Index

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5.10 H.323 Routing ....................................................................................................... 27 5.11 VoIP......................................................................................................................... 28 5.12 SIP Telephone........................................................................................................ 28 5.13 GIPS Voice Technology ......................................................................................... 29

Appendix: Server Port Information ................................................................................... 30

PeopleLink videoconference system Technical Whitepaper Chapter 1 Product Introduction

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Chapter 1 Product Introduction

PeopleLink enables users to have conferences via the Internet with a personal computer,

standard video capturing devices, speakers, and microphone. Compared with traditional

hardware based video conferencing system, PeopleLink provides users a much lower cost but

with a high quality, reliable, and efficient audio/video communication tool that has other

functions such as file/document sharing, and conference management.

The PeopleLink videoconference system comprises the PeopleLink Server, PeopleLink

Management System, and PeopleLink Client.

The setup requires one or more conference server installed with the server software and

given a fix public IP Address. If the conference servers are installed behind a firewall, TCP

ports 5222, 8082, 8083, 8099, and 18080 must be opened.

PeopleLink Server System Requirement is as follow:

Minimum Requirement Recommended Requirement

CPU Pentium IV2.0GHZ Pentium IV3.0G or above Dual

XEON CPU

Graphic card Integrated graphics card Independent graphics card

Memory 1Gb 2Gb to 16Gb depending on no of

concurrent licenses

Supported Recommended

Operating System Windows 2000/2003/XP/Vista/7

Windows 2003 Server

Besides the conference servers that need to be installed, all client computers that would be

connecting to the conference servers would also need to install the appropriate client

software in order to communicate with the server. The specifications for these client

computers are as below:

Minimum Requirement Recommended Requirement

CPU Pentium III 800 Pentium IV 2.0G and above

1G and above

Graphic card Integrated graphics card Independent graphics card

Memory 256MB 1G and above

Supported Recommended

Operating System Windows 2000/2003/XP/Vista/7 Windows 2000/XP/Vista

Multimedia Devices

PeopleLink videoconference system Technical Whitepaper Chapter 2 System Functionalities

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Chapter 2 System Functionalities

2.1 System Highlights

PeopleLink was developed based on IETF’s Extensible Messaging and Presence Protocol (XMPP) standard. It is a distributed, modular, and powerful software based

enterprise videoconference platform. The system takes into account expandability

and extensions of functions, allowing for users to experience instant communication

with each other, hold online conferences, provide conference notifications, etc. This

provides the user with a complete and unified network multimedia communication

solution. The PeopleLink system supports the master-slave server multi-level cascading

deployment, allowing the CPU and network processing bottleneck to be distributed to

multi servers. In such way, several thousand users may videoconference at the same

time.

Audio and video transmission would greatly improve in a network cluster architecture.

In this architecture, users are able have a better audio and video experience, this

also allows the subsidiary servers to smoothly connect to the system, thereby greatly

reducing the burden of the client, at the same time, these servers are also able to

disconnect without affecting the overall operation of the whole system. This audio

and video cluster architecture allows for integration with the email client and other

hardware videoconference terminals, this not only remove differences between the

systems, but it has also provided users with a great way of communication, cutting

down on spending and maintaining the value of their system, allowing users to keep

their past investments.

The Active/Active mode is supported, with mirror simulation software being used to

enable storage sharing. The mirror partition for the primary and secondary servers

must be a partition with the same size and volume number. The downtime of the

primary server is detected by checking the communications between the primary and

secondary servers. When the primary server fails, the secondary server takes over its

work and becomes the primary server. And the former primary server is re-started

and becomes the secondary machine. Only a dongle is needed for the primary and

secondary servers, with the dongle being attached to the primary server.

In order to satisfy the need for large number of users to be part of the conference session, the videoconference system supports videoconference multicasting. This

feature allows the use of minimal network bandwidth to be received by a big number

of users, which is the most effective way of internal multimedia communication.

The system also allows for multiple conference sessions, this means that on the same system, there can be multiple conference sessions with different resource

requirements. Each of these conference sessions are independent and do not disturb

other sessions. The system with its stable system structure and advance capabilities


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helps to ensure long periods of conference reliability. The videoconferencing system is cross platform compatible. It can be run on

Windows, Linux and UNIX operating system platforms.

The videoconference management system uses standard ODBC database interfaces

which support MySQL, Oracle, and other databases, and according to requirements

can be expanded as needed.

The system also has support for future SIP communication. Taking into consideration, communication support for current and future videoconference systems, the system

through implementation of a XMPP/SIP gateway, is able to do so. The great

expansion capability of the XMPP protocol has further given the videoconferencing

system greater expandability support.

System supports mobile client. You just need to have an smart phone supporting 3G or WIFI network (Android or ios system) and access to meeting via 3G or WIFI

network anytime and anywhere, avoiding drawbacks that you can not attend

teleconference until you turn on computer. You can see and hear, by means of mobile

phone, distal video images and voice communication. Meanwhile, your voice and

video can be conveyed back to meeting site. Moreover, you can look up documents

and materials being discussed in meeting room through mobile phone.

Using the Internet Explorer web browser, users can download, install, or upgrade components of the PeopleLink Client as needed. Given PeopleLink System’s user

friendly interface, users would find it easy to use and navigate around the software.

The system is fully web based. When the user enters the conference session, the system would check and determine whether the latest version of the client software

has been installed, if not, then it would automatically prompt the user to begin

installation. The installation files have been digitally signed and verified by VeriSign.

Through this way, this ensures the security of the client software and would prevent

infection by computer viruses and Trojans.

If you are RTX user, you just need to install simple plug-in modules to access to RTX. Even though establishing communication system and signing in meeting room, you

also can invite other good friends to attend your own meeting.

The system supports software smart updates. This means that when the user enters the videoconference system, the smart update feature is able to automatically verify

what components that needs to be updated on the client’s computer. This way, this

would minimize the bandwidth used for data transfer and simplifies the upgrade

process, further increasing the user friendliness of the system. The smart update

feature support retransmission, this means that if a previous update was not

completed, during the next update, the system would continue from where it last

stopped.

The system provides a testing conference environment for users to test their local and remote video display, check their network connections and make changes to

their audio/video configuration. This test conference session is created when the

Master MCU is started up and would always remain there, it does not occupy any

system resources.


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The server integrating and splitting function allows to integrate multiple conferencingsystems into a huge conferencing system. The working principle is to convert a

former master server into a slave server. After being registered on the central master

server, the local encrypted information needs to be reported to the central master

server, which centrally manages all the reported encrypted resources, forming a

conferencing system that supports more participants and conference rooms; if

necessary, the slave server may also be converted into a master server, allowing the

system to get rid of the control of the former conferencing system and become a

stand alone conferencing system. With popularity of the conferencing system,

resources sharing requirements on multi conferencing systems have been put on top

of the agenda. With the integrating and splitting function, resources of multi

conferencing systems are expected to be managed and shared efficiently.

2.2 Audio Video Highlights

The system utilizes G.723, G.711 and GIPS audio compression algorithms, this helps

to ensure stable audio clarity under all network bandwidth conditions. Coupled

together with the company’s patented audio improvement algorithm, this helps to

ensure that audio quality would not significant deteriorate under the worst network

conditions.

This system also utilizes the industry’s leading audio mixing techniques, which caneasily control voice switchovers.

The system could display the speaker’s audio pitch and also make adjustments toother user’s audio volume.

By introducing the H.263 and H.264 video technologies, the H.263-based video

communication between the standard client and H.323 video endpoint may be

conducted. Compared with H.261, H.263 only uses half of the bandwidth to enable

conferencing call between the standard client and H.323 video endpoint, with the

same video quality to be achieved. Featuring flexibility and bandwidth and storage

space saving, H.263 not only provides users with high quality video experiences but

efficiently uses the bandwidth resources, thus protecting users’ existing investment.

Videoconferencing between standard clients can also be performed under the H.264

mode. H.264 has very high coding efficiency, in the same image quality, it can save

about 50 percent of the bit rate than H.263. The bitstream network structure of

H.264 has strong adaptability, and increases the error recovery capabilities, it also

can adequately meet IP and wireless networking applications. Including H.264 VBRvariable bandwidth and H.264 CBR stable bandwidth.

Our system features a full range of high quality audiovisual resolutions: in videocodec applications, our system features H.264 codec algorithm; in video processing

applications, our system integrates the patented video pre-treatment,

post-treatment and displaying algorithms, which, by integrating with video

compression/decompression algorithms, not only efficiently eliminate video

distortion incurring at the video capture and encoding/decoding stages, but

considerably improve video quality and reduce the bandwidth used by video; in lip


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synchronization applications, the patented lip synchronization algorithm is

introduced to the system that also incorporates the cutting-edge GIPS audio and

video technologies, allowing excellent lip synchronization effects available over the

internet.

The system has good network adaptability and video compression rates suitable for

56K dial-ups to broadband environment, coupled with the company’s patented data

flow and network adaptability mechanisms, this guarantees that during bad network

conditions, the videoconference system would still be able to provide good video

display. If supported by encrypted device, the videoconference system supports high definition video with resolution up to 1920*1080. The frame rates of the video display

corresponds accordingly with the video transmission rates,

The system supports multiple remote video channels, this gives the users the ability to select, switch channels as they wish. The resolution and layout of the video

displays can be done by the system or done manually by the user, they can choose to

focus, do full screen enlargement, multiple displays, etc. to animatedly arrange the

layout.

The system supports two kinds of display ratio, 16:9 and 4:3, users can easily switch between them according to their needs, and make video window match the ratio of

video sources to achieve the best display effect.

The system supports video window banner title. Users can add banner title to their own video window to indicate the meeting topic and meeting agenda. The privilege of

determining whether to enable this banner title function is specific to the conference

moderator.

Based on the network conditions and requirements, the user is able to change the

bandwidth requirements and videoconference mode, thus enabling all levels of

videoconferencing.

Support the dual displays of video and data contents in the conference session. It

also has the ability to support remote webcam control, settings adjustments, etc.

2.3 Data Collaboration Features

Users are able to use the virtual whiteboard to write down or edit information. The virtual whiteboard has the ability to support multiple users’ interaction, this is a great

convenience for all conference users to communicate with each other. The use of a

virtual highlighter, laser marker, etc. further enhances the convenience and flexibility

of it.

The system provides the function of document sharing including normal documents sharing, dynamic PPT files sharing and web co-browsing. The file sharing ability allow

printable documents to be displayed on the virtual whiteboard, giving all conference

users a view of these files, at the same time, it also enables multiple users to work on

it at the same time. This sharing function also allows IE supported contents and

audio/video contents to be shared. The system supports sharing of multiple

documents, fully ensuring the utilization of this capability.


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Under the direction of someone, the conference users can use the online web sharing capability to browse websites and web pages. It supports simultaneously browsing of

multiple websites.

A media file can be used by the user as virtual equipment source which plays

audiovisual information contained in the media file in the conference.

The system supports TV wall function. Users can output multi-channel video signal to different TV sub-screens through the installation of specific decoder card to watch the

images of captured video on television screen conveniently and can easily switch

channels.

The system also supports desktop sharing, this means that the conference presenters can conveniently share their desktop with conference users to

demonstrate actual processes and applications to them, which is ideal and

convenient for collaboration and training purposes. When given the permission, the

users can even allow their desktops to be controlled and operated by another remote

user. Desktop sharing allows users the convenience of remotely controlled operations,

providing them with a new experience.

If display device supports touch, you just need to slide your fingertip gently in window7 system to realize such functions as mobile browsing, front/rear pages

turning, zoom in/zoom out etc in electronic whiteboard and shared document

contents. You can also move window at will to the position you require in floating

video window mode.

The user can do screen capture of any part of the screen, and use that captured still image to be displayed in a new virtual whiteboard.

The system also provides standard text messaging and file transfer capability. All

conference users are able to use the common text messaging platform to message

each other, they also have the able to have private chats with a specific user as well.

During the conference session, using the file transfer function, a file can be

conveniently transferred to a specific conference user or to everyone in the

conference session. In the File Transfer Management screen, the local user is able

manage uploading or downloading of files. The Conference Moderator can also

instantaneously clear all file transfers.

The presentation controller (the person who is hosting the conferencing session) has the right to present files/documents, build new whiteboards, etc., he can also do

co-browsing with all other users in the conference.

PeopleLink allows sharing of applications or the whole desktop during the conference

session. The shared applications or desktop can be remotely controlled by the other

user who has access control over it.

PeopleLink allows for both public and private text chat communication and file transfer.

The system provides the ability to record conference sessions, the user can record the contents of the conference session, and save it to the local hard drive.

The conference moderator has the ability to create conference agenda and after having access as the presentation controller, run the conference agenda. When the


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conference has entered a particular agenda, the related document would appear in

the presentation area.

The conference scheduler and administrator can set up the conference evaluation function. It can enhance the communication between conference managers and

users. After the user exit the conference, a evaluation page will appeared, user can

feedback information on it.

The system administrator has the rights to issue the system announcements. User can view the published announcements in the login page to master the latest system

information speedily.

The system administrator can send system broadcast to all users to enable user to

receive the real-time system information conveniently.

2.4 Conference Management

The videoconference system is based on the online web approach, this mean that

remote conference management is possible, management functions includes, user

management, conference scheduling, conference configurations, conference

notification, conference cancellation or extension, and managing conference server

resources and queries, etc.

The system supports multiple levels of user management, the whole organizational structure is displayed in the system in a branch directory tree model. The highest

administrator of the system, which is the Admin user, can start or configure the

system.

There are several ways of authenticating the user in a conference session. According to the different security levels needs, when scheduling a conference session, it can be

done based on several options: 1) Whether allow guest users. 2) Whether allow

monitor users. 3) Determine whether a conference password is needed. 4)

Determine whether conference users must be registered users. 5) Whether to hide

the conference ID. 6) Whether to encrypt the data in the conference session. It is

based on these options that different levels of security authentication can be offered

to the user.

The conference moderator can set the conference session to be either in free mode or moderator mode. In free mode, speech, access controls are managed in a queue

system. In moderator mode, the conference moderator has centralized over all

functions and can manage the whole progress of the conference session.

Conference system can combine the monitor system, administrator can configure the

monitor functions when the server supports monitor commutate service. During the

conference, moderator call the monitor user, and then display its video in the

conference. Monitor user has no other rights as other common conference user.

In the conference, user list can show as tree corresponding with the department frame of management system, it can also flat display, and users can sort the list

according to the whether top, function priority, department order, user post level and

user order.


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The conference session supports guest user, which is enabled by the hardware encryption device. Guest users are unable to upload audio/video, they can only

receive conference information indirectly. When scheduling a conference session,

access controls can be set for these users.

The PeopleLink supports periodic confer schedule, the system would schedule the

next meeting automatically at the end of each conference according to the

conference recurrence parameter. It would provide a great convenience for the

periodic confer schedule and simplifies the administrator’s schedule process.

When scheduling a conference session, it must be done according to the resources given, this includes conference time, no. of participants, bandwidth, video displays,

etc. Being able to manage conference session resources this allows for the control of

network bandwidth utilization when sending video. This prevents the problems

caused by over utilization of the network bandwidth used in order to send video.

The system supports the use of email for conference notifications, users just need to

click on a URL link embedded in the email to directly enter the conference session.

The system also supports the use of SMS for conference notification, the conference scheduler just need to send a conference notification to the cell phone no. of the user

in the organization.

System provide two kinds of Conference resources using mode for user to improve the utilization rate of conference resources and to enable the important meetings

holding stably.

The system supports automatic alerting function. In this function, the system administrator can monitor the system. If an exception happens, the administrator

can find out the reason quickly and take immediate action.

The system supports roll call function. The moderator can call over the names of

participants on the roll-call list which be set up in the management system by

scheduler.

2.5 Connection Capability

There are several solutions that the system can use when connecting through the NAT, proxy servers and firewall, and it can still send multimedia contents using HTTP.

This has to a large extent helped to overcome communication problems posed by

firewalls, this has freed many organizations of the need to change any network

configuration, which enable easy deployment of the system to the existing network

environment.

The system support Real-time Transmission Protocol (RTP) for multimedia data transfer, this has to a large extend help latency caused by an unstable network.

PeopleLink provides various solutions for bypassing the NAT, proxy server, and several types of firewalls. With HTTP Tunneling technology, most of the firewall

communication problems can be solved without changing the current network

configuration. At the same time through use of anti jittering techniques, packet loss

mechanisms, network adaptation algorithms and the company’s patented streaming

control technology, this has help greatly minimized the communication problems


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caused by the network environment, guaranteeing decent quality in a complicated

network situation. The system adopts a centralized network resource approach, this helps to ensure

that network resources are used in a controlled manner, efficiently and effectively,

this is to prevent the danger of excessive or improper use of network bandwidth. The

QoS used by the system supports IPLR, DiffServ and other mechanisms.

The user can connect to the videoconference through many different ways, thisincludes dialup, ISDN, ADSL, LAN, HFC, DDN, and other broadband connections.

In the system’s distributed architecture, with servers located at different locations,sometimes there can be short disconnections, and during these short disconnections,

this can have serious consequences to the videoconferencing system. In this system,

such disruptions would automatically cause the system to try to reconnect to the

main server, in the meantime, the user can select to connect from other servers to

enter the conference session or choose to wait. If the user chooses to wait, once the

server is back online, the user would automatically reenter the conference session.

In a multiple server environment, the user can select a server from which to connectfrom. This is why the videoconference system provides for an Intelligent Routing

capability, which arranges each server based on their connectivity. Usually, user

would select to connect from the server that has the best connectivity, this can help

ensure a great conference experience.

The communication between the client and server are encrypted, this ensures thatsensitive information can be securely transferred over the network. According to

different types of data, the system utilized the following encryption algorithm:

Diffie-Hellman algorithm (RSA 1024 characters), DES algorithm (RC4 128 Bit),

Message-Digest algorithm (MD5 128 Bit).

PeopleLink videoconference system Technical Whitepaper Chapter 3 Conference Server System Structure

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Chapter 3 Conference Server System Structure

PeopleLink is a distributed web-based conferencing system, which is a group of servers

(connected together in a star network topology) connected together, this means that there is

a main central server connected by multiple other servers. All data processing and network

management tasks are shared these servers, which enable the system to simultaneously

conferences of up to a few thousand people. The system is up as long as the network

connection is maintained from the subsidiary server to the central server; this organizational

structure would increase the load capacity of system and expand the whole system capacity.

Connecting the servers requires a central controlled and managed approach, through the use

of the central server for configuration and management purposes, this enables active control

and management of components of all subsidiary servers. With the ability to remotely

troubleshoot problem and manage these servers, there isn’t a need to for someone to take

charge of the server room.

The conference system takes into account the unique problems faced by small and medium

enterprises (SME). Through the use of the NAT and other methods by the subsidiary server to

connect to the Internet, this means that in small branches that do not have their own unique

public IP address, these branches are also able to setup a subsidiary server and have the

ability to connect as well.

The diagram below shows the videoconference’s system structure:

Figure 3-1


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3.1 JabberD

XMPP (Extensible Messaging and Presence Protocol) created by the Jabber community is used by PeopleLink servers for all its communication signaling needs. JabberD acts as the nerve

center for all communications signals sent by user to the system as well as internal

communication within components in the system. Because it was designed to handle multiple

user connections, even a standalone JabberD server can support up to a million simultaneous

connections. Through the system’s use of multiple Client to Server (C2S) components, this

can support even bigger volume of user connections.

3.2 MCU

The MCU (Multipoint Control Unit) primarily manages the conference sessions and the signaling from these conference sessions. All users that enter or exit the conference session,

that includes all access rights given to the conference user are managed by the MCU.

3.3 Multimedia Server

The multimedia server of the system is to receive, manage and transmit the audio and video

data that it received. Since it is audio and video that users are most concerned about in a

conference session, so the largest amount of processing and network resources are given the

multimedia server.

3.4 File Server

File and document sharing is the system’s most important data collaboration function. With the server, the user can either do a virtual print of their documents or do web sharing, which

would allow other conference users to immediately view this information. Through the use of

the file server, this also enables real time document transfer between conference users.

3.5 Whiteboard Server

The whiteboard server is responsible for managing, verifying and transmitting the contents of

the conference session’s “virtual” whiteboard. Through the combined use of the “virtual”

whiteboard and file sharing capabilities, conference users can work together on a file, which

allows for convenient data collaboration.


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3.6 Application Sharing (AppSharing) Server

In some conference session, the user might want to a share particular application or even an entire desktop with the rest in the conference, or even to allow another user to take remote

control of the desktop or application. This is handled by the Application Sharing server which

accepts, processes and transmits the necessary signals.

3.7 Conference Management Server

The conference management systems provides for database configurations, server startup

configurations and control capabilities. Database configurations mainly include initialization

of the database and updating the current database. Server configuration includes local and

remote adding of a server, generating the configuration file, which with the generated

configuration file start the services of the local or remote server and also to monitor the

operational status of services.

In most cases, many conference sessions must go through the scheduling process. The

conference management system can help the user to conveniently schedule a conference,

manage and query it. The system administrator through this system manages all users and

conference sessions. The conference management system is based on JSP, databases and

web services, the user just need to use an Internet browser to go to the relevant webpage,

enter the user name and password to access it.

3.8 C2S Module (Optional)

By configuring multiple C2S modules, this would enable the even distribution of online

connections to each module, each C2S module maximum supported online user capacity is 1

million, 5 C2S modules would be able to handle a system of 5 million online connections.

All server components cn, according to the needs can be configured on the same computer or

on a different computer, and according to requirements, configure multiple similar server

modules on one system. Usually, the multimedia server and the application sharing server

require a lot of bandwidth and processing resources, so it is a priority to consider increasing

these two servers and distributing them onto different computers, this is to guarantee system

stability and capacity.

The image below is an example of a distributed system configuration, the system has

configured two multimedia servers, which are operating on separate different computers.

There are two application sharing servers as well on two different computers, one server that

operates the conference management system and database and another server operating

the MCU, file server, whiteboard server and JabberD.


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Figure 3-2

PeopleLink videoconference system Technical Whitepaper Chapter 4 Communication Standards

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Chapter 4 Communication Standards

4.1 Communication Protocol: IETF XMPP

The Extensible Messaging and Presence Protocol (XMPP) is the IETF's formalization of core

protocols used to expand communication and online conferencing. The work on XMPP was

based on Jabber and Jabber was soon a solution for instant messaging systems, it was then

put on the Internet as an open source organization and continued to expand from there. As

opposed to other instant messaging system protocols, there are four main aspect of XMPP: it

is XML bases, it is a distributed network system, it is based on open standards, and the

system design is highly modular and has an expandable structure.

Because of the fact that XMPP was broadly designed based on email technology, it is similar

to the email system, which has become the Internet’s most successful communication tool.

Different XMPP systems which pass through a XMPP protocol can spontaneously form a

distributed system, through this system, different client applications are able to freely

transmit messages to other clients, and it does not take into account if both parties are on the

same server.

The difference between XMPP and the email system storage/transmission approach is that

XMPP has the ability to send almost real time information, this is because the server is able

to know if a particular user is online, this is know as Presence in the XMPP system. The XMPP

system also has a unique feature, which makes it the leading standard in system messaging.

First of all, it is based on open standards, which enabled it to be the communication bridge for

different messaging systems, secondly, with XMPP strict adherence to the XML standard, its

highly modular structure, together its intelligent message transmission capability, all of these

have given users the ability to communicate with each other, but it also made communication

between different applications possible as well.

4.1.1 Distributed Network

The XMPP network architectural structure and the email system are similar, with each user requiring a local server to receive and send messages. Servers can exchange information and

online messages with each other as well. Because of that, there can be multiple servers at

any given time on the network, with each server performing operating independently and

also having a user list of their own as well. As long as any two servers are able to query each

other, messages can be exchanged as well. Because of the fact that the user account is

related to the server, the user ID looks similar to an email address, e.g. [email protected]


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Figure 4-1

4.1.2 C/S Structure

The XMPP utilizes a Client/Server architecture, this means that all XMPP information and

messages must pass through the server before it can reach other users. Whether there may

be direct data communication links between clients, these solutions still relate to the overall

concept as well, in fact, the negotiation process to setup these links goes through the XMPP

server in the first place.

4.1.3 Server Modularity

There are two main purposes of the XMPP server: Receive client’s connection request and

directly communicate with the client application, and to communicate with other XMPP

servers as well.

The design of the XMPP server full incorporate the concept of modular design, where by

different coding modules handle different functions, e.g. user authentication, data storage,

etc. At the same time, the server can expand its capabilities by adding new servers,

centralized control for safety management, adding external server components, etc.

With communication modules as an example, communication between the XMPP system and

other systems is completed by the Transport module. This component is responsible for

converting the XML information sent by the XMPP into the protocols of other related system.

This module is not part of the XMPP core module, but an external add on component to the

server, this dynamically provides extra functions to the system.


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4.1.4 XML Format

The two great features about XML is its flexibility and expandability, this enables it to display

information in any forms, because of this ability to do so, XML became the core XMPP system.

In reality, on the XMPP system, whether it is a connection from the client to the server, the

communication link between servers, or on the server’s configuration file, XML is everywhere.

One example:

<message from='[email protected]' to='[email protected]'> <body>Wherefore art thou, Romeo?</body> </message>

4.2 Audio/Video Transmission Protocol: RTP

In order to ensure real time transmission of multimedia content, the videoconference system

utilizes Real-time Transport Protocol (RTP) to transfer audio and video data, this has to a

large degree minimized the effects caused by latency in an unstable network situation. The

company’s own anti jittering algorithm can handle jittering and incorrect queuing of data

packets. The system also uses the latest packet loss recovery mechanism, which significantly

increase its ability against network dropping of data packets. Even in a relatively bad network

situation, with high packet loss, the system is still able to provide high quality audio service.

Besides that the system also utilizes static noise reduction, echo cancellation, audio boost

and other techniques, all of these help to minimize network utilization but increase the audio

quality as well.

4.3 Audio/Video Transmission Protocol: HTTP

With the increasing emphasis by users on network security, there are an increasing number

of users that are implementing network firewalls or NAT and other related security

mechanisms. All of these network security mechanisms help to ensure the network security

of the network of the business organizations, but this has brought many difficulties for

videoconferencing systems. Standard H.323 and SIP systems utilize RTP protocol for audio/

video communication, but RTP is unable to pass through the firewall or NAT.

Firewall and NAT are more and more being deployed when connecting to the Internet due to

various security purposes and this brings a problem for web conferencing. This is because

web conferencing typically uses H.323 or SIP protocols, together with RTP for audio-video

transmission but unfortunately the RTP protocol does not work well in a firewall/NAT

connected network environment.

When deployed in a firewall or NAT environment, standard H.323 and SIP videoconferencing

systems would usually utilized the below methods:


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1) Convince the client not to use the firewall or NAT. This may sound funny, but in reality

many clients accepts this suggestion due various reasons.

2) Install the system in the Demilitarized Zone (DMZ). This meant that the system is

installed outside of the corporate network, in a location without the protection of the

firewall or NAT. With this approach, even it the system was attacked, the internal network

of the company would not be affected at all.

3) Utilizing network tunneling, application layer devices, proxy servers or other solutions.

These are not the standard components of the videoconferencing system, but in order to

safely ensure the coexistence of both the videoconferencing system and the firewall, it

would be the only way.

The above 3 solutions, one of it sacrifices the safety of the videoconferencing system, another

increases the complexity of the system and also increases the investment costs of the client

as well. From the client’s point of view, these are not great solutions.

The PeopleLink system supports the use of RTP, and at the same time, through the

implementation of standard HTTP protocol to enable audio and video transmission. This

means that no matter how strictly regulated the firewall is, it would still allow HTTP protocol

access to pass through it as one of the basic communication protocol, or else this would mean

total disconnection of the network with the outside world. The videoconference system uses

HTTP to send out audio and video data, this means that as long as anyone is able to access

the Internet, videoconferencing would be available to them as well. Clients would be able to

use the videoconferencing system without having to increase their investments or sacrifice

network security. Suffice to say, this is the client’s best choice.

PeopleLink videoconference system Technical Whitepaper Chapter 5 Main Communication Standards

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Chapter 5 Main Communication Standards

5.1.1 H.263, H264 Series of Standards

H.263 is a provisional ITU-T standard. It was designed for low bitrate communication. But, in

fact, this standard may be used for a wide range of bitrates, not just low bitrate applications.

It is expected that H.263 will replace H.261 in many applications. The coding algorithm of

H.263 is similar to that used by H.261, however with some improvements and changes to

improve performance and error recovery. Being used in low bitrate applications, H.263

provides better picture quality than H.261. The differences between them are listed below. 1.

Half pixel precision is used for motion compensation whereas H.261 used full pixel precision

and a loop filter; 2. Some parts of the hierarchical structure of the data stream are now

optional, so the codec can be configured for a lower data rate or better error recovery; 3.

There are now four optional negotiable options included to improve performance; 4.

Unrestricted Motion Vectors, Syntax-based arithmetic coding; 5. Advance prediction, and

forward and backward frame prediction similar to MPEG called P-B frames; 6. H.263 supports

five resolutions. In addition to QCIF and CIF that were supported by H.261 there is SQCIF,

4CIF, and 16CIF. SQCIF is approximately half the resolution of QCIF. 4CIF and 16CIF are 4 and

16 times the resolution of CIF respectively. Essentially, H.263 has replaced H.261.

H.264 is developed by the ITU-T VCEG (Video Coding Expert Group) and Joint Video Group

(JVT: joint video team) of ISO / IEC MPEG (Moving Picture Expert Group) as a new digital

video coding standard, it is both ITU-T H.264, and also Part 10 of ISO / IEC MPEG-4.

H.264 and previous standards are also mixed coding mode mixed by DPCM and transform

coding. It adopted "return to the basic" simple design, do not have many options, the

compression performance was much better than H.263 + + . It strengthened the adaptation

capacity of various channels, and adopted the structure and syntax of "network friendly", it is

beneficial for error handling and packet loss; wide range of objectives to meet the different

rates, different resolutions and different transmission (storage) occasions demand.

Technically, the H.264 standard has many advantages, such as the uniform VLC symbol

coding, high-precision, multi-modes displacement estimates, based on the 4 × 4 integer

transform, hierarchical coding syntax, and so on. H.264VBR/CBR has very high coding

efficiency, in the same image quality, it can save about 50 percent of the bit rate than H.263.

The bitstream network structure of H.264VBR/CBR has strong adaptability, and increases the

error recovery capabilities, it also can adequately meet IP and wireless networking

applications.

5.1.2 Reason for using H.264 VBR/CBR

High Compression: Compared with the same series of compression standards like MPEG1 and MPEG2, or ITU’s H.261, H.263 standards, H.264VBR/CBR still has a clear


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advantage when dealing with video compression rates. This is important when it can

effectively save precious bandwidth in videoconferencing. Bandwidth Adaptability: H.264VBR/CBR is not only suitable for use in low bandwidth

(less than 64 Kbps) environments but it is flexible enough to be used by a high quality

and bandwidth connection, satisfying the different demands of the users. The users can

configure the bandwidth according to their requirements, PeopleLink would then fully

utilized the bandwidth given and provide a satisfying videoconferencing experience.

5.1.3 Future of Video Compression

With the constant development of network technologies, there is no doubt H.264VBR/CBR is

State of the Art video compression algorithms and beats any other Compression technology

in the world.

5.2 Audio Codec: G.723.1

International Telecommunication United (ITU) G series audio compression standards

comparison chart.

Codec Type Rate (Kbps) Time Delayed (ms)

G.711 A-Law / µ-Law 64 0

G.722 SB-ADPCM 64/56/48 0

G.723.1 MP-MLQ/ACELP 6.3/5.3 37.5

G.726 ADPCM 16/24/32/40 0

G.727 Embedded ADPCM 16/24/32/40 0

G.728 LD-CELP 16 < 2

G.729 CS-ACELP 8 15

5.2.1 G.723.1 Audio Quality

According to international standards, Mean Opinion Score (MOS) is a commonly used method

for evaluating the quality of human speech at the end of a circuit. Based on MOS results, the

audio quality of a telephone call has a score of 4 (good). This means that the user should not

be able to tell the difference from a telephone call to a compressed audio output that has a

MOS score of 4.

The PeopleLink system utilizes G.723.1 audio compression, its compressed audio has a MOS

score of 3.98 (Source: DSP Group Inc., http://www.ds pg.com), which is very close to a

telephone call quality.


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5.2.2 Why use G.723.1 for audio compression

a) High Compression Rate: Compared with other similar compression standards, G.723.1 has the highest compression rate (up to 6.3 Kbps) among all of them.

b) Good Audio Quality: After compression, a G.723.1 compressed audio has a MOS score of 3.98, which to most untrained ears, isn’t able to distinguish whether or not the audio has

been compressed. It is this audio quality that can completely satisfy the needs of online

web conferencing.

5.3 Packet Lost Reduction Algorithm

There are several factors that can affect the Internet’s Quality of Services (Qos), as shown

below:

a) Internet Bandwidth: This is refers to the rate of data that can be sent every second. The effect of broadband services is easy to see, the higher the bandwidth, the more data gets

transmitted, which means even better quality and effects. But bandwidth is not the only

determining factor.

b) Packet Queuing: There are many nodes in the network, e.g. router, gateways, etc. All of

these nodes used a packet queuing mechanism to determine the order of sending the

data packets. If at a certain node, there is a sudden long queue of data packets, this

node would then start dropping data packets to ensure that the node is functioning

normally. Even if these packets are not dropped, after a long period in the queue, these

packets would take a very long time to reach their destination, this would cause network

delays and jittering. Packet dropping, latency, extended jittering would have a big impact

on videoconferencing. c) Data Classification: Just like RTSP and IPv6 and other standards, data are classified

before being sent. Because of the higher requirements of these protocols, equipments

must be upgraded or reconfigured again, this has resulted in a slow adoption of these

protocols.

One on hand, as broadband is now more and more widely used, this means that more and

more customers would have access to higher end broadband connection. On the other hand,

the quality of broadband connections differs very much. In a lot of cases, the client may have

high end broadband connection, but at the same time there is also a high rate of packet loss

and severe network jittering, all this these would have severe impact on the quality of

videoconferencing, particularly audio quality.

It the company’s aim to enable client to be able to hold videoconferencing at the present

network environment, this requires that the company’s videoconferencing system to be able

to smoothly adapt to the different qualities of current existing networks and not just require

that the client provides high quality connections.


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After many years of testing and research, the company has managed to develop a multimedia

packet loss algorithm. The enables the videoconferencing system to have packet loss rate of

up to 10% in the network environment and still be able to guarantee audio continuity and

good audio quality. Having a decent overall audio quality is a prerequisite of having a success

videoconference session. After using this new patented technique, the company has

successfully allowed hundreds of client companies to hold videoconferencing sessions on the

Internet, with good results to show.

5.4 Bandwidth Adaptability

During a conference session, there may be times when there can be frequent network

fluctuations. If these network fluctuations are too great, this may cause abnormal audio/

video transmission in the videoconferencing session, this would lead to audio and video

degradation in the conference session. Usually under these situations, the audio would sound

broken and video display would turn blur. This is when the videoconference system has to

make the necessary corrections to make up for the network fluctuations.

After having done numerous investigations at the client’s place, we discovered that when a

network fluctuation happens, the user’s acceptable level of video would have a slight

degradation, but the requirement for continuous and clear audio quality is still there. Because

of that, the PeopleLink videoconferencing was designed that when it detect that there is a

change in network bandwidth, it would first of all, lower the video frame rate in order to

guarantee audio continuity. The frame rate change mechanism is that when it detects that the

network bandwidth is lower what the system requires, it would lower the video frame rate to

half of the original. For example: if the original frame rate is set at 20 fps, when the system

detects insufficient network bandwidth, it would lower the frame rate to 10 fps. If the system

detects that it is still not enough, it would go even lower to 5 fps. When the system finally

detects that performance is back to normal, it would slowly adjust the frame rates upwards,

from 10 fps to 11 fps, 12 fps to 13 fps …… until it stabilized to the nearest network bandwidth

value. During the whole process, the audio data rates would still remain unchanged.

5.5 Video Frame Rate

In a conference session, the bandwidth for each conference user may not be the same. Many

videoconference systems can only offer low bandwidth conference services, when broadband

users connects to the conference session, they would also receive low bandwidth quality as

well. The PeopleLink videoconference system allows for many frame rates to exist together in

one conference session.

Before the conference session begins, each client would first check the bandwidth connection

with the server, and notify the server. If the server detects different bandwidths on the end

connections of the conference session, it would then handle the audio/video data separately.


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For broadband users, the system would send broadband quality data, for low bandwidth users,

the server would make some modifications to lower the data quality in order to suit the

network requirements and send it to the user. It is because of the existence of such a

mechanism, that the PeopleLink videoconference system can maximize the user of network

bandwidth, letting users experience the best quality in videoconferencing.

5.6 Desktop/Application Sharing Bandwidth Adaptability

Desktop/application sharing is a very practical function, by using this function, the user is

able to remotely view or even control the operations of the remote computer. There are two

factors that affect the practicality of the function. On one hand, the network resource

requirement for this function is higher and on the other hand, when used in reality, even in

the same conference session, different user would usually have different network connection.

How does a user under a low bandwidth situation be able to use desktop sharing? How can a

broadband user and a low bandwidth user be able to see the same desktop sharing view in

respective of their bandwidth connection?

The videoconference system, through its own special bandwidth adapting mechanism can

send the data to the users based on their bandwidth. When the user does not have enough

bandwidth, the server would automatically lower the refresh frame rates of the desktop

sharing process and send it to the user, if there is enough bandwidth for the user, the server

would then increase the refresh rate of the sharing process and send it faster to the user. It

is through this bandwidth adapting mechanism, that even when a 56Kbps dialup user and a

2Mbps dedicated leased line user are in the same conference session, the server is still able

to guarantee the dialup user normal use and at the same time, allow the dedicated leased line

user to view a higher quality of the desktop sharing process.

5.7 High Quality Broadband Video

With the more widespread use of broadband, users are now demanding for better and higher

quality of audio and video. To address these demands, PeopleLink videoconferencing

software has focused on the broadband environment and optimized the video and audio

quality for it.

In a broadband environment, even at 512Kbps bandwidth where actual bandwidth will be

average 200Kbps downlink and 80-100Kbps uplink the Video Quality will be quite Smooth and

even though the bandwidth falls below 30-40Kbps the Video call will not get disconnected as

other VC Solutions but only it will dynamically reduce FPS and maintain small Video Quality

and audio quality.


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5.8 Hardware Selection

During the design stage of the videoconference system, the designers have taken into

account compatibility with various different hardware devices. Because of the fact that the

system itself is totally hardware based, it through the selection of external hardware devices

that enables the videoconference system to provide a low level to high level

videoconferencing solution.

5.8.1 Desktop/Laptop Setup

When talking about setting up the desktop end, we need to take into consideration, costs and

portability. It is suggested that the user uses a microphone and a webcam that supports USB

connections, the user would just need to plug in the webcam and microphone directly into the

desktop computer or the laptop and immediately sign on to the videoconference system to do

videoconferencing. For those who are always on the go, this setup is the most convenient of

all.

5.8.2 Conference Room Setup

The audio/video requirements when setting up in a conference room are usually high, we

would suggest the user to utilize relatively high end equipment to connect to the system. For

more information, please take a look at the following chart:

Use Devices

Video Device

Webcam/PTZ Camera (e.g. PeopleLink i5, sg 501 PTZ, HD 380

ETC)

Video Card (PeopleLink USB Capture card or PCIe HDMI Capture

card)

Video Display Computer monitor/ projector/LCD/LED/LFD/VIDEO WALL

( through VGA, RCA, HDMI)

Audio

Recording Microphone

Audio

Playback Speaker


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5.9 Communication with Videoconferencing Hardware

There are some clients who already have their own videoconferencing systems. These

systems are typically total hardware based solutions, using ISDN (H.320) or IP (H.323) to

hold videoconference sessions. When these clients wish to further improve their systems,

they would usually have these two requests:

1) Even lower development cost. The cost of building a brand new videoconferencing

system would usually cost even more than the existing system. Even then, while still

using a hardware solution, the costs involved would still be very high, so it must be an

even lower cost solution.

2) Able to connect and communicate with existing hardware systems. The use of the

videoconference system is for communication, so when the users are using it, they

would expect that all videoconference systems would be able to connect and

communicate with each other and not operating independently alone, becoming a

closed system.

So in order to address these needs, the designers of the PeopleLink videoconference system

had come out with a solution that would allow it to connect and communicate with other

hardware based videoconferencing solutions, as shown below:

Figure 5-1

5.10 H.323 Routing

Video Multi-channel Multiplexing Technology Video stream carries a huge amount of data,

whereas, in most cases, network bandwidth resources are limited. More intensifying video


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data streams may interrupt the continuity of video stream.

The video multiplexing technology allows multiple video streams to be smartly combined for

transmission over one existing video channel. In this way, video data is transmitted in an

organized way, making video data transmission become more efficient. It helps to optimally

use network bandwidth resources and makes data transmission become smoother.

5.11 VoIP

PeopleLink is dedicated to providing a full set of enterprise communication solutions to high

quality videoconferencing applications and video telephone/Ethernet telephone access and

management applications, thus capable of providing integrated communication solutions

including point-to-point audiovisual calling and multi-point videoconferencing; our solutions

are based on H.323 protocol. This means any device that supports this protocol may use our

solutions and may be managed by PeopleLink videoconferencing system; our system

features robust and flexible user management, access validation, conference routing, call

management, and number allocation. As well as centralized video/Ethernet telephone

management, allowing fully integrated multi-media communications to be implemented.

The videoconference system provides a PSTN gateway which can enable normal telephones

or mobile phones to seamlessly connect to the IP based conferencing system. This means

that the user can use the telephone to connect into the videoconference system and chat with

anyone in the conference session. In fact the conference moderator is able to invite a

particular telephone caller to enter the conference session through the use of the system’s

voice system.

The system also provides for multiple simultaneous call connections, this would completely

satisfy large organization’s conference needs and at the same time increases the choices for

the users as well, this mean that users who do not have access to a PC are able to use a

normal telephone or mobile phone to join a conference session, and be able to talk in it as well.

The system’s powerful conference management feature allows a telephone user to easily

communicate with other telephone callers or normal conference users in the conference

session at the same time, it is also able to let the user talk to a specific person, or have a

discussion within a smaller group.

The PSTN gateway uses the RTP protocol to connect to the IP based videoconferencing system and the PSTN network, it supports ADPCM、G.711 PCM、PCM, OKI ADPCM and many

other audio codec, which when combined together with the company’s patented audio

optimization algorithm, this would greatly increased the system’s audio quality.

5.12 SIP Telephone

SIP, also called Session Initiation Protocol, was put forward by IETF (Internet Engineering

Task Force) in 1999. SIP is a signaling protocol that allows real time communication

applications to be implemented over IP networks, especially the Internet. This protocol is


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used in a wide range of applications such as VoIP and instant messaging. Also it will be used

as the core transmission protocol for NGN, making it a promising application-layer control

protocol.

PeopleLink is devoted to the creating a cutting-edge VC Solutions in the multi-media sector.

Our SIP-based videoconferencing soft-switch gateway is designed to cater for the intensifying

SIP users’ needs. Any device that supports the standard SIP protocol can be connected to the

PeopleLink videoconferencing system via this gateway; you may choose to perform access by

yourself or by receiving invitation from other privileged users; after entering the conference,

these SIP endpoints may participate in conferences just like other standard conferencing

terminals. They may speak, listen, receive/send video and perform other conferencing

functions. In this way, the applications of the PeopleLink videoconferencing system are

expanded. It also allows the SIP systems to be used in many new applications.

5.13 GIPS Voice Technology

1. International leading audio codec technology, allowing dynamic adjustment depending on

network bandwidth, thus providing excellent voice quality that traditional telephone doesn’t

have.

2. Real time automatic gain control (AGC) technology, allowing automatic volume adjustment

depending on speaker’s current state, thus ensuring comfortable listening experience for

participants. Meanwhile, it makes speaker’s voice become more natural.

3. Advanced echo cancellation (AEC) technology, efficiently eliminating echo interference

arising in the multi-party conference, thus ensuring a more clear voice.

4. Advanced noise reduction (NR) technology, allowing efficiently eliminating background

noise that may negatively affect speaker’s voice quality, thus providing more comfortable and

clear voice effects.

5. Real time voice active detect (VAD) function, allowing to efficiently reducing bandwidth use

for multi-party conference, thus improving multi-party voice quality.

6. Excellent multiple audio mixing technology, ensuring no voice distortion for multi-party

conference.

7. GIPS is a IETF - Standard Protocol which is a High Efficiency Variable Bit Rate Codec

mostly used by software based Audio & video Solutions.

PeopleLink videoconference system Technical Whitepaper Appendix: Server Port Information

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Appendix: Server Port Information

Services Listening ports

Protocol Direction Description

Apache 80 TCP In/Out Management system Tomcat 8080 Machine Loader 23459 TCP In gsoap communication between

ML and the management system C2S 5222 TCP In Connection between the server

and client. S2S 15269 TCP In Connection between the servers Master MG 8086 TCP In Master multi-media gateway

audio port 8087 TCP In Master multi-media gateway

video port Slave MG 8084 TCP In Slave multi-media gateway audio

port 8085 TCP In Slave multi-media gateway video

port 33000～50000 UDP In/Out Audiovisual port range of the

multi-media gateway under UDP mode

Master AppShare 5523 TCP In Master applications sharing data transmission port 5524 TCP In

Slave Appshare 5525 TCP In Slave applications sharing data transmission port 5526 TCP In

MasterFS 8099 TCP In File data uploading port of the master file server

SlaveFS 8100 TCP In File data uploading port of the slave file server

Http Tunnel 443 TCP In Tunnel service Machine Loader 23457 TCP In Remotely start or stop the server

23458 TCP In H323/VoIP 1720 TCP In Public H323/VoIP service port,

with dynamically distributing UDP port

7000 TCP In 1719 UDP In/Out 1718 UDP In/Out

SIP 5060 UDP In/Out Public SIP service port, with dynamically distributing UDP port

H323 Route 7000 TCP In Public H323 service port 1719 UDP In/Out 1718 UDP In/Out

PeopleLink videoconference system Technical Whitepaper Appendix: Server Port Information

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Services Listening ports

Protocol Direction Description

jabbergw 18083 TCP In Gsoap listening port (cannot be modified)

SlaveMG for h323

8088 TCP In Extended H323 multi-media gateway audio port

8089 TCP In Extended H323 multi-media gateway video port

33000～50000 UDP In/Out Audiovisual port range of the extended h323 multi-media gateway under UDP mode

SlaveMGfor pstn 8090 TCP In Extended pstn multi-media gateway audio port

8091 TCP In Extended pstn multi-media gateway video port

33000～50000 UDP In/Out Audiovisual port range of the extended pstn multi-media gateway under UDP mode

SlaveMG for sip 8092 TCP In Extended sip multi-media gateway audio port

8093 TCP In Extended sip multi-media gateway video port

33000～50000 UDP In/Out Audiovisual port range of the extended sip multi-media gateway under UDP mode

SlaveMG for voip

8094 TCP In Extended voip multi-media gateway audio port

8095 TCP In Extended voip multi-media gateway video port

33000～50000 UDP In/Out Audiovisual port range of the extended voip multi-media gateway under UDP mode

PPT Animation 10000 TCP In Port for PPT Animation

Note:

The above ports are defaulted ports. You may modify them when configuring.

If only part of the ports is allowed to be used on the client, the conferencing system orsome functions may not be used. This may only be solved through tunnel service. Forencrypted conferencing service, the tunnel service must use Port 443.

The MG’s UDP port rang is calculated like this: first configure the lowest UDP port, thehighest UDP port=the lowest UDP port+encrypted users×3. If UDP is used, all the portsbetween the lowest firewall UDP port and highest firewall UDP port need to be opened.

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