Thiết bị và công nghệ hỗ trợ Video, âm thanh audio cho các giải pháp Meeting Center–Sametime


Trước hết chúng ta phải tìm hiểu về Meeting Center của Sametime có cấu tạo như thế nào ?

Phần 1. Audio & Video trong Sametime 8.5.x

Introduction

This article

  • provides a Sametime 8.5 audio and video (A/V) guide, with links to useful references
  • summarizes features of Sametime 8.5 audio and video
  • shares some good practices and troubleshooting techniques
  • provides diagrams of A/V call flow to assist with planning and troubleshooting

This document is an accumulation of knowledge and lessons learned from many teams’ contributions – issues uncovered from IBM Software Services for Lotus (ISSL) customer engagements and architectural discussions, lessons learned from test cycles, and Sametime Development’s in depth knowledge of the component and its intended design.

Review system requirements

First of all, review the Lotus Sametime 8.5 Detailed System Requirements document to make sure you are aware of the requirements and limitations of the Sametime 8.5 components for audio and video (A/V).

Requirements important for A/V are as follows:

  • In the Sametime Media Manager Server section, note that Sametime 8.5.0 Media Manager is supported only on Windows and Linux platforms.
  • The notes for the Sametime Media Manager Server section are as follows:
    • Sametime Audio/Video Reflector (STUN/ICE server) is not supported
    • No IPV6 support
    • No FIPS support
    • WebSphere Application Server 7.0.0.3 Network Deployment (included with installation)
  • In the Audio / Video Client Requirements section, note the following:
    • Windows XP or Windows Vista operating systems
    • Memory requirements of a minimum of 128 MB of memory for video
    • Requires a Sametime Media Manager server for voice chat and video calls unless using Classic Meetings
    • Pre-Sametime 8.5 clients cannot participate in voice chat and video calls using the Sametime 8.5 Media Manager server
      • (Stated another way, this means all participants must be connected to the same community used by the Media Manager, and all participants must be using Sametime 8.5 Connect client for Audio and Video)
    • Default bitrate (CIF 352×288@15Fps 384kbps)
    • 6-person video limit in chat or meeting
    • 10-person audio limit in chat or meeting
    • For a positive audio video experience a network latency of no more than 100 milli-seconds is required between the clients and servers. In addition a network bandwidth of of 200-300 kbps is required per video stream between clients and servers. A network bandwidth of 30-40 kbps is required per audio stream. The network bandwidth requirements will vary depending on the quality of audio, video, and number of streams selected.
    • Sametime video codecs provide many resolution choices, from SQCIF to Wide Full HD (1080p). The higher the resolution, the more CPU, display memory, and graphics card power are required. Machines equivalent to Lenovo T60 can handle CIF and VGA, but HD will require Intel Core 2 Quad or better CPU and at least 256 megabytes of display memory.
  • In the Audio / Video Client Requirements section, the Web client is listed as not supported, except on the Classic Meetings on only Windows platform.
    • What this means is that A/V in Web meetings requires that such A/V Web Meetings be started through the Sametime 8.5 Connect client. Web A/V directly through a Web browser is not supported (namely, opening a browser, and typing in the URL directly to a Sametime Meeting Server). A Web A/V meeting must be entered through the Sametime Connect client user interface.


Check setup for A/V

For audio and video to work properly with Sametime 8.5 clients, ensure that you have the following installation and configuration in place:


Sametime 8.5 audio and video features

This section outlines what’s new in 8.5 audio and video features and changes from previous versions of Sametime.


Client changes

In Sametime 8.5, the point-to-point audio/video and n-way audio has been radically changed in both the stand-alone and embedded clients. A Media Manager (also known as Media Server) is now used to perform all types of A/V calls, even for point-to-point calls. The 8.5 client now uses SIP and TCSPI to establish A/V calls; the 8.5 client does not support legacy phone grid based audio/video.

As noted in client interoperability in the Information Center, all participants must use 8.5 clients for A/V features; there is no backward compatibility for A/V with older Sametime clients. Sametime 8.5 clients are not backward compatible with previous versions for audio and video. Even with Media Manager, 8.5 clients will not be able to place audio and video calls to older clients (including SUT 8.0 clients). This limitation in backward-compatibility is due to the major improvements introduced in 8.5 with new audio and video codecs and multi-way audio/video, including a change to SIP-based standards. Therefore, there is a one-time switch to the new technology.
When an 8.5 client interacts with a pre-8.5 client, you see the following behavior:

  • If the 8.5 client has access to Media Manager, a call invitation from an older client will be declined automatically
  • If the 8.5 client does not have access to Media Manager, then an older client might display an error message when trying to call an 8.5 client (depends on the older release)

If you are upgrading from Sametime 8.0.2 client to Sametime 8.5, then make sure the Media Manager is also installed. Without the Media Manager, there will be no computer device and UI controls to start audio or video calls; they will be disabled.
The Information Center topic “Managing media codecs” lists the supported codecs for Sametime 8.5, summarized as follows:

  • Video: H264, H263
  • Audio: ISAC, iLBC, G.722.1-32, G.722.1-24, G.722.1-16, G.711


Audio/Video feature comparison by version

The following table shows audio-visual features by the version which includes them:

Feature

8.0.2 without ICE

8.0.2 with ICE

8.5.0

8.5.2

Point-to-point (P2P) AV

Supported

Supported

Supported
Media Manager Server Component required
Non NAT

Supported
Media Manager Server Component required
NAT’ed Support a TURN Server is required

Multiway (n-way) AV

Supported

Supported

Supported
Media Manager Server Component required
Non NAT

Supported
Media Manager Server Component required
NAT’ed Support a TURN Server is required

NAT and traversal between heterogeneous networks crossing firewalls

Only via Reflector, has inconsistent performance

Uses ICE to support many configurations, including using Reflector/TURN server

Not supported

Supported
Media Manager Server Component required
NAT’ed Support a TURN Server is required

Backward- compatible

Yes

Yes

No

No

AV in online meetings
(using Connect Client)

Yes

Yes

Yes, using 8.5 Connect Client

Yes

AV in online meetings
(browser, not using RCP)

Yes
(Classic Meeting Server MRC)

Yes
(Classic Meeting Server MRC)

AV in online *new* Sametime 8.5 Meeting Server requires 8.5 Connect Client

Classic Sametime 8.5 server still features AV in Java-based online meetings

Yes, Requires a Sametime Media Server and Sametime Proxy Server Compontent.


Making and joining calls

There are multiple ways for users to make or join a call or video call in Sametime 8.5, such as using the telephony icon or using the “Call” or “Start Video call” menu options from the Contact list. For steps and sample screen captures, refer to the following resources:
Voice and video with Lotus Sametime software
Lotus Sametime Help > What other ways can I communicate beyond text chats?

Meeting integration

When combined with the Lotus Sametime Media Manager, meeting rooms can be enhanced with audio-visual features.

  • Using audio and video in chat (not Sametime Meetings) for point-to-point and multi-point does not require a meeting server. A/V is integrated with meetings when using the Sametime client.
  • For audio/video to work in meetings, the Meeting Server and Media Manager are both required.
  • Meeting rooms can be created with A/V or, in the case of an instant meeting, A/V can be added later to the room.

By default, only six participants with video are supported in a meeting room. To change the setting, refer to “Limiting participants in a video conference.”
For steps and sample screen captures of using audio-visual features in meetings, refer to the following resources:
Online meetings with Lotus Sametime software
Lotus Sametime Help > Make conference calls and use video

Third-party service provider support

Starting with Sametime 8.5, TCSPI has been extended to support video. Previous third-party TCSPI service providers can still be used with the Sametime 8.5 Community Server, but to do so telephony service needs to enabled. All new TCSPI third-party service providers must be deployed on the Media Manager. At this time, only one external provider being in use is supported. The external provider can use its own SIP based MCU to switch RTP data between Sametime Connect clients and different SIP endpoints. A sample “MyAV” TCSPI adapter is included in the 8.5 Software Developer Kit (SDK). The following diagram, an excerpt from the SDK, shows this sample MyAV adapter integrated to Sametime 8.5:
MyAV sample service provider topology

About the Media Manager

This section describes the Sametime Media Manager, including its architecture, components, and installation options. The Sametime Media Manager allows users to start point-to-point and multipoint A/V calls with each other using the Connect client and meetings.

Architecture

The Sametime 8.5 Media Manager consists of three server components:

  • SIP Proxy and Registrar
  • Conference manager
  • Packet switcher

For details on each component, refer to the Information Center topic “Lotus Sametime Media Manager.”

  • The Proxy and Registrar handles endpoint registration and SIP-based call routing.
  • The Conference manager hosts TCSPI adapters (one internal and one external) and provides conference focus functions (as in RFC 4353).
  • The Packet switcher routes audio and video to participants based on voice-activated switching. Note that point-to-point A/V calls do not use the Packet switcher.

All Media Manager components are WebSphere-based applications, and they must be running for audio and video to work. These components can be on a single machine or distributed across multiple WebSphere nodes.
This diagram shows the components of the Media Manager and their port ranges:
Media Manager port ranges
The diagram shows all the Media Manager’s WebSphere applications running in a single server deployment of the Media Manager.
Media Manager applications

Proxy and Registrar

The SIP Proxy and Registrar contains and provides a list of all registered components and clients. It loads server configurations from proxy.xml and registrar.xml files. All SIP messages pass through the SIP proxy and are routed to the registered endpoints. There can be no SIP communication if the SIP Proxy and Registrar server is down. You can install multiple SIP Proxy and Registrar components and cluster them for high availability and failover.

Conference Manager

The Conference Manager provides call control, management, and notifications. The Conference Manager also starts the Sametime Telephony service on start-up. It loads internal and external TCSPI service provider using server configurations. It loads server configurations from the stavconfig.xml file in WebSphere Application Server. The Conference Manager registers with the SIP Proxy and Registrar that it is running. Clients get notified of the Telephony service and TCSPI service provider via VP protocol. Clients then register with the SIP Proxy and Registrar. During conference calls, the Conference Manager acts as a back-to-back user agent (B2BUA) for the client.

Packet switcher

The Packet switcher is responsible for receiving and sending media streams from endpoints to other endpoints in a conference. The Switcher works on audio streams to determine the active video stream to send to the participants, a process known as Voice-Activated Switching (VAS).
The Packet switcher loads server configurations from the stavconfig.xml file in WebSphere Application Server and registers with the SIP Proxy/Registrar. It periodically sends availability messages to the Conference Manager. In an A/V conference call, it acts as a SIP endpoint.
It is important to note that the Packet switcher dynamically allocates ports to all audio and video conferences and reuses these ports once A/V calls are ended.
Clustering of Packet switcher is not supported, but there can be multiple packet switchers on different hosts. If you have a Conference Manager cluster, then the Packet switcher is registered with the Conference Manager cluster, and each cluster member uses the same Packet switcher.

Installation options

You install the Lotus Sametime Media Manager using the Sametime System Console (SSC). The SSC can manage only one Media Manager server. (While it is possible to install without using the SSC, that configuration is not advised for production environments.)
You can use the “All components” deployment plan to install a stand-alone Media Manager in proof-of-concept environments.

If you are clustering Media Manager, take note of the following items:

  • Clustering requires proper planning
  • Only the Conference Manager and SIP Proxy and Registrar can be clustered
  • The Packet switcher cannot be clustered, but there can be multiple switchers
  • The “All components” installation (Proof of concept/Pilot installation option) cannot be used for clustering
  • At least three machines are required for vertical clustering where each component uses the SSC deployment manager:
    1. 1 x Conference Manager: Network Deployment – Primary node
    2. 1 x Proxy and Registrar: Network Deployment – Primary node
    3. 1 x Packet Switcher: Network Deployment – Primary node

Additional Packet Switchers can be deployed on different machines

    WebSphere Application Server Proxy server can be used on the same machines

Steps can be found in Sametime 8.5 Information Center topic “Clustering Lotus Sametime Media Manager components.”


A/V in Sametime Classic Meetings

The Domino-based Sametime 8.5 Classic Meeting Server allows for online Web meetings using the Sametime Meeting Room Client (MRC). This Java-based Web client allows for A/V to be used in online meetings. It interacts with the Sametime Classic Meeting Server; it does not interact with the new features in the 8.5 client, nor the Media Manager (and its components), nor the Sametime reflector.

The audio-visual features of the Sametime Classic Meetings client are:

  • All A/V packets relayed through the Sametime Classic Server
  • Audio supports the G.723, G.711 and iSAC codecs
  • Video only supports the H.263+ codec
  • Neither the client or server supports NAT traversal, but the server does have the option to tunnel AV packets over TCP.

Only a limited number of H.263 video resolutions are supported for this classic client, namely only the following H.263 CIF formats are supported:

  • CIF, 352x288QCIF, 176×144
  • SQCIF, 128×96

Newer webcams might not support lower resolutions. The Classic Meeting Client does not prevent the user from choosing an unsupported format. As a result, video will not appear even though audio continues to function properly. There is no anticipated change for the Classic Meeting Client to support another codec or additional formats, so users are encouraged to try using different webcams to resolve the problem.


Best practices

Review good practices in the following resources:

Practices that are important to highlight:

  • Use a supported camera, and use the camera manufacturer’s drivers instead of the generic Windows drivers
    List of supported cameras from System Requirements: http://gipscorp.com/support/ibm
  • All participants must be connected to the same community used by the Media Manager. All participants must be using Sametime 8.5 clients.
  • One of the key factors affecting video quality is available network bandwidth. The higher the video resolution, the more bandwidth is required for better quality. Adjust and manage video bit-rates accordingly on the Media Manager Video Resolution configuration page.


Troubleshooting resources

General troubleshooting tips can be found in the Lotus Sametime Wiki article “Troubleshooting the Sametime 8.5 Media Server.”

If you need to collect diagnostic data (logs and trace files) to review, either on your own or when working with IBM Support, refer to “Collecting Data: Lotus Sametime Media Manager.”


Known issues and field notes

This section highlights some known issues and troubleshooting notes. You can also search technotes about known issues or frequently asked question in the IBM Support Portal. Or browse through Sametime 8.5 technotes on audio/video.

Problem: Poor video quality

Solution: Video quality is a combination of the end user’s camera as well as the compression level set on the server + network bandwidth and latency available to support it.

Test the camera performance outside of Sametime to check whether Sametime is a factor. For example, you might notice that a laptop-integrated camera has poorer quality than an external camera even outside the Sametime environment.

There have been issues with cameras not working with Sametime if the Windows default drivers are used. The camera might test fine in Windows, but the Sametime client does not recognize the camera or show it in the preferences. To resolve, use the camera-specific drivers. If in doubt about whether the drivers are a factor, the UIM logs are a source to check for data.

Problem: Getting ‘ST_CONNECT_NOT_PRIVILEGED’ or ‘ST_CONNECT_HOST_UNREACHABLE’ error in media server logs.

Solution: Add the media server IP in the trusted IP list of the ST community server. For more details see technote 1429413 and Information Center topic “Configuring a Lotus Sametime Community Server to allow connections from Conference Manager nodes”

Problem: AV does not work for n-way and meetings call, only point-to-point AV calls work.

Solution: All n-way and meetings AV calls require that at least one Packet Switcher is available to the Conference Manager.

Typically this problem occurs due to configuration settings

  • Verify that a PacketSwitcher.info file is present in the media server logs directory
  • Open the stavconfig.xml file under the DM profile directory in WebSphere Application Server and verify that the value of packet switcher host is the same as the machine on which it is running
  • Might need to remove any host aliases from the OS hosts file
  • Media Manager must be deployed on the primary host name of the machine if there are multiple IP/hostnames available on the machine
  • Verify that none of the clients and media server are behind a NAT firewall
  • Clients and media server should be able to open connections (or ping) in both directions
  • UDP must be allowed through firewalls. Test by disabling firewalls on client and server and verify A/V
  • Diagnose the media server logs and enable tracing if necessary

Problem: UDP port conflicts

Solution: Identify if UDP ports are already being used. If Sametime 8.5 finds that UDP port is being used, or there is a conflict, it will not try to use it again. Refer to the instructions from the following Sametime Information Center topics to see if you need to modify the UDP ranges:

Modifying the dynamic port range to improve Packet Switcher performance

Managing UDP ports for voice chat and video calls

Problem: How to modify the maximum number of participants in an A/V session?

Solution: By default, only six participants with video are supported in a meeting room. To change the setting, refer to “Limiting participants in a video conference.”

Problem: Some participants in a meeting room might see “waiting for call to start….” though the moderator has started the A/V call already

Solution: A hotfix is available; see technote “Meeting participant sees Waiting For Moderator in an instant meeting room” (#1427723).


Call flows and interactions

This section provides diagrams of various call flows or configurations to help you plan or troubleshoot in your environment.


Call flow for a point-to-point AV call

The following diagram shows how two Sametime 8.5 clients handle a point-to-point A/V call. Client A starts a call with Client B by sending TCSPI requests to Conference Manager through the Sametime Community server. The conference manager creates a new conference and communicates with both clients over SIP through the SIP Proxy/Registrar. Finally the clients negotiate the media and start sending audio and video data directly to each other.

Figure 9.1 – Point-to-point call
Point-to-point call


Call flow for a multi-point AV call and AV meetings

The following diagram shows how three or more Sametime 8.5 clients handle a multi-point A/V call. Note that the same sequence also applies to A/V calls in Sametime meetings where there can be one or more clients. Client A starts a call with Client B and Client C by sending TCSPI requests to Conference Manager through the Sametime Community server. The conference manager creates a new conference and communicates with the Packet Switcher and all three clients over SIP through the SIP Proxy/Registrar. All the clients negotiate the media with the Packet Switcher and start sending audio-video data to it. The Packet Switcher sends the data back to all participants in the call using voice-activated switching.

Figure 9.2 – Multi-point call
Multi-point call

Point-to-point interception

In this diagram, endpoint A starts a point-to-point call with endpoint B by sending TCSPI requests to the conference manager. Both endpoints are then added to the conference by the conference manager and receive a SIP INVITE request. After getting SIP ACK from the conference manager, the endpoints are able to send/receive media between each other.
Figure 9.3 – Point-to-point interception
Point-to-point


Escalation from point-to-point to multi-point

This diagram shows how a point-to-point A/V call between endpoints A and B is promoted to a multi-point call by adding endpoint C. The steps are similar when a multi-point call is started directly. The new endpoint is added to the existing conference by Conference manager using TCSPI and sent a SIP INVITE request. Endpoint C then negotiates media with the Packet Switcher and start sending/receiving media data from it.

Figure 9.4 – Escalation to multi-point
Multi-point


Media Manager Installer and System Console interaction

The diagram depicts the steps involved when the Media Manager is installed using a deployment plan created in the Sametime System Console. All of the Media Manager configurations are captured by the installer and stored in the stavconfig.xml file, which is used by the Conference Manager and the Packet Switcher components of the Media Manager.

Figure 9.5 – System console
System console


Media Manager admin flow

In this diagram, the SSC pushes all the Media Manager’s configurations and client policies on to the SSC database, which are then pulled by the Community server and used by the Sametime 8.5 client. The Community server will synchronize any configuration or policy changes made on the SSC server using the default time interval of 60 minutes.
Figure 9.6 – Administration flow diagram
Administration flow diagram

Conclusion

You can achieve a successful implementation of audio/video features in Lotus Sametime 8.5 by understanding the components, checking requirements, and planning. If problems do occur, review best practices, troubleshooting resources, and call flow diagrams.

 

Phần 2. Thiết bị hỗ trợ chất lượng âm thanh tốt

Tham khảo: http://www.clearone.com/personal-usb-speakerphone.html

Có nhiều dòng sản phẩm, nên chọn với không gian của phòng họp để phù hợp:

ví dụ: 50 người họp

Personal USB speakerphone delivers unmatched audio and connects to multiple devices

The CHAT® 50 personal speaker phone is a mobile audio peripheral that connects to a wide variety of devices and provides crystal-clear, hands-free audio communications.

It provides unmatched full-duplex capability, which allows users to simultaneously speak and listen without audio cutting in and out. It also provides high-quality audio playback for music, gaming and other sound files.

The CHAT 50 can be used in a variety of ways, and connects to the following devices:

Personal USB Speakerphone

Laptops, PCs, or Macs — for use with:

  • Internet telephones, such as Skype or Vonage
  • VoIP softphones, such as Avaya, Cisco, Nortel
  • Web conferencing applications, such as LiveMeeting, Sametime, and WebEx. View our Web Collaboration Partners
  • Instant messaging with audio chat
  • Audio playback with media players, such as QuickTime, RealPlayer, Windows Media Player
  • PC-based gaming with TeamSpeak

Telephones* — for hands-free, full-duplex conversations
Cell phones* — for hands-free, full-duplex conversations
Desktop video conferencing devices — for hands-free, full-duplex conversations Desktop Videoconferencing Partners
iPods & MP3 players — for full-band audio playback (single speaker)

*Note: connectivity to telephones and cell phones is limited to specific models. Cables that connect to the RJ-9 headset jack of certain Avaya, Cisco, and Inter-Tel models

Hoặc thiết bị cần Full-duplex audio cho cả đường hình kết nối:

The CHAT 150 USB connects to PCs for rich, full-duplex audio communications, and can be used with VoIP softphones, web collaboration applications, instant messaging, and any other application requiring two-way audio.

Connects to laptops and PCs — for use with:

  • VoIP softphones, such as Avaya, Cisco, Nortel
  • Internet telephones, such as Skype or Vonage
  • Web conferencing applications, such as LiveMeeting, Sametime, and WebEx. View our Web Collaboration Partners
  • Instant messaging with audio chat
  • Audio playback with media players, such as QuickTime, RealPlayer, Windows Media Player

USB speakerphone

Featuring ClearOne’s trademark audio processing performance, CHAT 150 delivers unmatched full-duplex capability for crystal-clear communications. It includes three built-in microphones for full 360-degree pickup.

The CHAT 150 is a perfect addition to the office or conference room for greatly enhanced collaboration.

The CHAT 150 also connects to the following devices:
Enterprise telephone handsets* — for hands-free, full-duplex conversations
Video conferencing systems — for hands-free, full-duplex conversations Desktop Videoconferencing Partners

*Note: Connectivity to telephone headsets is limited to specific models due to variations in the headset jack to which the CHAT 150 connects. The CHAT 150 Avaya has been tested for compatibility with Avaya 2400, 4600, 9600 series phone models. The CHAT 150 Cisco has been tested with Cisco 7940, 7960, and 7970 phone models.

 

m ClearOne — see part number list. Also, many home and business telephones feature a 2.5mm headset jack; the CHAT 50 USB Plus comes with a 2.5mm-3.5mm cable which connects to these phones as well as most cell phones.

 

Phần 3: Các cổng port được mở trong Sametime:

IBM® Sametime uses a number of ports on the servers in your deployment. This topic lists the default ports and their uses; a range of ports means that the application can select any port in that range, in case one or more of those ports are already in use by other applications.


Sametime System Console

The following ports are used on the Sametime® System Console.

Table 1. Sametime System Console ports

    Default Port
    Purpose
    50000
    Installation manager utilities, post-registration utilities, and the Sametime Meeting Server access the Sametime System Console database port. The database port number is determined by the DB2® server configuration.
    9080
    The Sametime Community Server accesses the Sametime System Console HTTP port. This is determined by the WebSphere® Application Server configuration. You can find this port number in AboutThisProfile.txt or in the Integrated Solutions Console.
    9443
    The Sametime Community Server accesses the Sametime System Console HTTPS port. This is determined by the WebSphere Application Server configuration. You can find this port number in AboutThisProfile.txt or in the Integrated Solutions Console.
    8700
    Provides HTTP browser access to the Sametime System Console for administrators. This is determined by the WebSphere Application Server configuration. You can find this port number in AboutThisProfile.txt or in the Integrated Solutions Console.
    8701
    Provides HTTPS browser access to the Sametime System Console for administrators. This is determined by the WebSphere Application Server configuration. You can find this port number in AboutThisProfile.txt or in the Integrated Solutions Console.

DB2 server

The following ports are used on the DB2 server.

Table 2. DB2 server ports

    Default Port
    Purpose
    50000
    The DB2 port is accessed by the Sametime System Console. The port number is configured by the DB2 server configuration.

LDAP server

The following ports are used on the LDAP server.

Table 3. LDAP server ports

    Default Port
    Purpose
    389 or 636
    The LDAP port is accessed by the Sametime System Console. The port number is configured by the LDAP server configuration.

Sametime Community Server

The following ports are used on the Sametime Community Server. The first table lists ports used by HTTP Services, Domino® Services, LDAP Services, and Sametime intraserver ports, and the second table lists ports used by Community Services.

Table 4. HTTP Services, Domino Services, LDAP Services, and Sametime intraserver ports

    Default Port
    Purpose
    80
    The Sametime Community Server listens for the Sametime System Console on port 80.

    If you allow HTTP tunneling on port 80 during the Sametime Community Server installation, the Community Services multiplexer on the Sametime Community Server listens for HTTP connections from web browsers, and Sametime Connect clients on port 80.

    If you do not allow HTTP tunneling on port 80 during the Sametime Community Server installation, the Domino HTTP server listens for HTTP connections on this port.

    Alternate HTTP port (8088)
    If you allow HTTP tunneling on port 80 during the Sametime Community Server installation or afterward, the Domino HTTP server on which the Sametime Community Server is installed must listen for HTTP connections on a port other than port 80. The Sametime installation changes the Domino HTTP port from port 80 to port 8088 if the administrator allows HTTP tunneling on port 80 during a Sametime Community Server installation.

    Note: If you allow HTTP tunneling on port 80 during the Sametime Community Server installation, web browsers make HTTP connections to the Community Services multiplexer on port 80, and the Community Services multiplexer makes an intraserver connection to the Sametime HTTP server on port 8088 on behalf of the web browser.

    This configuration enables the Sametime Community Server to support HTTP tunneling on port 80 by default following the server installation.

    389
    If you configure the Sametime Community Server to connect to an LDAP server, the Sametime Community Server connects to the LDAP server on this port.
    443
    The Domino HTTP server listens for HTTPS connections from the Sametime System Console on this port by default.

    This port is used only if you have set up the Domino HTTP server to use Secure Sockets Layer (SSL) for web browser connections.

    1352
    The Domino server on which Sametime is installed listens for connections from Notes® clients and Domino servers on this port.
    9092
    The Event Server port on the Sametime Community Server is used for intraserver connections between Sametime components. Make sure that this port is not used by other applications on the server.
    9094
    The Token Server port on the Sametime Community Server is used for intraserver connections between Sametime components.

Table 5. Community Services ports

    Default Port
    Purpose
    1516
    Community Services listens for direct TCP/IP connections from the Community Services of other Sametime Community Servers on this port. If you have installed multiple Sametime Community servers, this port must be open for presence, chat, and other Community Services data to pass between the servers.
    1533
    The Community Services listen for direct TCP/IP connections and HTTP-tunneled connections from the Community Services clients (such as Sametime Connect and Sametime Meeting Room clients) on this port.

    Note: The term “direct” TCP/IP connection means that the Sametime client uses a unique Sametime protocol over TCP/IP to establish a connection with the Community Services.

    The Community Services also listen for HTTPS connections from the Community Services clients on this port by default. The Community Services clients attempt HTTPS connections when accessing the Sametime Community Server through an HTTPS proxy server.

    If you do not allow HTTP tunneling on port 80 during the Sametime installation, the Community Services clients attempt HTTP-tunneled connections to the Community Services on port 1533 by default.

    80
    If the you allow HTTP tunneling on port 80 during the Sametime Community Server installation, the Community Services clients can make HTTP-tunneled connections to the Community Services multiplexer on port 80.

    Note: When HTTP tunneling on port 80 is allowed during the Sametime installation, the Community Services multiplexer listens for HTTP-tunneled connections on both port 80 and port 1533. The Community Services multiplexer simultaneously listens for direct TCP/IP connections on port 1533.

    8082
    When HTTP tunneling support is enabled, the Community Services clients can make HTTP-tunneled connections to the Community Services multiplexer on port 8082 by default. Community Services clients can make HTTP-tunneled connections on both ports 80 and 8082 by default.

    Port 8082 ensures backward compatibility with previous Sametime releases. In previous releases, Sametime clients made HTTP-tunneled connections to the Community Services only on port 8082. If a Sametime Connect client from a previous Sametime release attempts an HTTP-tunneled connection to a Sametime Community Server, the client might attempt this connection on port 8082.

Table 6. Sametime Classic Meetings

    Default Port
    Purpose
    1533,
    The Sametime Classic Meeting Room client loads in a user’s web browser when the user attends an instant or scheduled meeting. The Meeting Room client must establish connections with the Community Services on the Sametime Community Server (on default port 1533).
    8081,
    The Meeting Room client must establish connections with the Meeting Services on the Sametime Community Server (on default port 8081).
    554
    The Sametime Classic Recorded Meeting client attempts a direct RTSP TCP/IP connection to the Recorded Meeting Broadcast Services on the Sametime Community Server on default port 554. Over this connection, the Broadcast client negotiates with the server to receive the streams that transmit the recorded meeting data.

Sametime Media Manager

The following ports are used on the Sametime Media Manager.

Table 7. Media Manager ports

    Default Port
    Purpose
    9080
    HTTP port for control and general management of audio/video calls. In a cluster, HTTP ports are proxied through a WebSphere Proxy Server. This lets you open these ports only between the firewall and the WebSphere Proxy. WebSphere may change this ports depending on the install environment.
    42000-43000
    The Packet Switcher component of the Sametime Media Manager routes audio data to participant endpoints through a range of ports starting with 42000 through 43000. It uses values in this range as needed, as it services multiple calls. It chooses new ports in increments of 2.

    If encryption is enabled (SRTP), the range starts with an odd port number. RTCP starts with the next port available, which is the RTP or SRTP port incremented by 1.

    46000-47000
    The Packet Switcher component of the Sametime Media Manager routes video data to participant endpoints through a range of ports starting with 46000 through 47000. It uses values in this range as needed, as it services multiple calls. It chooses new ports in increments of 2.

    If encryption is enabled (SRTP), the range starts with an odd port number. RTCP starts with the next port available, which is the RTP or SRTP port incremented by 1.

    5060 and 5061
    The Conference Manager, and Packet Switcher are SIP applications, so they use WebSphere SIP container ports. By default, they are 5060 and 5061, but they are dependent on WebSphere during install to determine the available port numbers to use. In a cluster, SIP ports are proxied through a WebSphere Proxy Server. This lets you open these ports only between the firewall and the WebSphere Proxy.
    8880
    This is for server to server communication. The Sametime System Console accesses the Deployment Manager SOAP port. This port number varies, depending on how WebSphere was installed. The port number can be determined by looking at AboutThisProfile.txt’s SOAP connector port value in the profile log directory or the Integrated Solutions Console.

SIP Proxy and Registrar

The following ports are used on the SIP Proxy and Registrar.

Table 8. SIP Proxy and Registrar ports

    Default Port
    Purpose
    5080-5081
    SIP messaging uses these ports in a single server Media Manager deployment where SIP Proxy and Registrar runs in a separate virtual host. The value is defined in the WebSphere Application Server instance on which the Sametime Proxy & Registrar is running.
    5060-5061
    The default ProxyRegistrar installer does not use these ports. It uses the two above. Therefore, this is only true if the administrator changes the virtual host to use the default, which is defined on port 5060/5061. SIP messaging uses this port in a multiple server Media Manager deployment where SIP Proxy and Registrar runs in on a separate machine. The value is defined in the WebSphere Application Server instance on which the Sametime Proxy & Registrar is running.

Sametime Meeting Server

The following ports are used on the Sametime Meeting Server. Most of these ports are configurable.

Table 9. Meeting Server ports

    Default Port
    Purpose
    9080
    In a single node environment using HTTP that bypasses the WebSphere Application Server proxy, the Sametime Meeting Server listens for data from the Sametime Meeting Room client over this connection.
    443
    In a single node environment using HTTPS that bypasses the WebSphere Application Server proxy, the Sametime Meeting Server listens for data from the Sametime Meeting Room client over this connection.
    9080
    In a multiple node environment using HTTP, the Sametime Meeting Server listens for data from the Sametime Meeting Room client that is passed through the WebSphere Application Server proxy.
    9443
    In a multiple node environment using HTTPS, the Sametime Meeting Server listens for data from the Sametime Meeting Room client that is passed through the WebSphere Application Server proxy.
    8880
    This is for server to server communication. The Sametime System Console accesses the Deployment Manager SOAP port. This port number varies, depending on how WebSphere was installed. The port number can be determined by looking at AboutThisProfile.txt’s SOAP connector port value in the profile log directory or the Integrated Solutions Console. See also the following section: Note about SOAP ports for complex deployments.

Note about SOAP ports for complex deployments Deploying WebSphere Application Server SOAP port is complicated and might include ports besides 8880, especially if there is more than one Sametime product on a specific machine. Usually the firewall openings are configured prior to the deployment, when your understanding of the port configuration is still incomplete. In order to have a smoother deployment you can add port ranges – for example 8880 – 8890 and 8600 – 8610. For example, when a Sametime Proxy node in the DMZ is federated into the internal Sametime System Console cell, it needs one port for Sametime System Console to Sametime Proxy Deployment Manager communication and another port for the Sametime Proxy primary node communication – which is on the same machine. You might also need port 8601 when you want to update the Sametime Proxy configuration through the Sametime System Console.


Sametime Proxy Server

The following ports are used on the Sametime Proxy Server.

Table 10. Proxy Server ports

    Default Port
    Purpose
    8880
    This is for server-to-server communication. The Sametime System Console accesses the Deployment Manager SOAP port. This port number varies, depending on how WebSphere was installed. The port number can be determined by looking at AboutThisProfile.txt’s SOAP connector port value in the profile log directory or the Integrated Solutions Console. See also the previous section: Note about SOAP ports for complex deployments.

Sametime Advanced

The following ports are used on Sametime Advanced. Most of these ports are configurable.

Table 11. Advanced server ports

    Default Port
    Purpose
    9080
    The default http port for the Sametime Advanced web application.
    9443
    The default https port for the Sametime Advanced web application.
    1883
    The default MQTT port. The broadcast community alerts and notifications are sent over this port.
    8990
    The default MQTT SSL port. The broadcast community alerts and notifications are sent over this port.

Note about SOAP ports for complex deployments Deploying WebSphere Application Server SOAP port is complicated and might include ports besides 8880, especially if there is more than one Sametime product on a specific machine. Usually the firewall openings are configured prior to the deployment, when your understanding of the port configuration is still incomplete. In order to have a smoother deployment you can add port ranges – for example 8880 – 8890 and 8600 – 8610. For example, when a Sametime Proxy node in the DMZ is federated into the internal Sametime System Console cell, it needs one port for Sametime System Console to Sametime Proxy Deployment Manager communication and another port for the Sametime Proxy primary node communication – which is on the same machine. You might also need port 8601 when you want to update the Sametime Proxy configuration through the Sametime System Console.


Sametime Connect client

The following ports are used on the Sametime Connect client.

Table 12. Sametime Connect client ports

    Default Port
    Purpose
    80
    The client listens for HTTP traffic over this port. This cannot be configured in preferences.
    22222
    The installed meeting client uses this port for peer-to-peer application sharing.
    20830 to 20930
    This range of ports is used by the audio and video channels to receive RTP and RTCP packets over UDP.
    5060
    Sametime Connect client SIP port. The Sametime Connect client will start with the initial port value, finding the first port available in increments of 2. This search is up to and including the starting port value plus 100. The starting value is implemented as a preference, but is not currently exposed for update via any user interface.
    5656.
    Sametime Connect client port used for peer-to-peer file transfer.
    59449
    Sametime Connect client Web API port for HTTP
    59669
    Sametime Connect client Web API port for HTTPS

 

Phần 4: Một số lưu ý khi cấu hình Sametime cho Audio/Video:

Managing UDP ports for voice chat and video calls

You can change the UDP ports for computer-to-computer voice chats and video calls.

About this task

IBM® Lotus® Sametime® comes with voice chat. With voice chat, users can place and receive audio calls with up to six participants by default, using their computer’s and their chat partners’ computer audio capabilities. Once a user has a computer-to-computer voice chat started, the user can convert it to a video call so that the user can both see and hear call participants.

Voice chat works with user datagram protocol (UDP) packets which flow through UDP ports on the firewall of every client machine to allow users to speak to other users orally over the computer. The client machines use a single port (UDP port 20830 is the default) for all audio chats, so this port must be opened for both incoming and outgoing UDP traffic.

Video calls also work with user datagram protocol (UDP) packets which flow through UDP ports on the firewall of every client machine to allow users to see video of users with whom they are chatting over the computer. The client machines use a single port (UDP port 20832 is the default) for all video calls, so this port must be opened for both incoming and outgoing UDP traffic.

Note: The client might require ports for the audio and video channels to send RTP and RTCP packets over UDP.

Follow these steps to change the UDP ports:

Procedure

  1. Log in to the Integrated Solutions Console.
  2. Click Sametime System Console > Sametime Servers > Sametime Media Manager.
  3. In the Sametime Media Managers list, click the deployment name of the Lotus Sametime Media Manager.
  4. Click the Configuration tab.
  5. The Lotus Sametime Media Manager listens for inbound audio streams from clients on a range of 100 UDP port numbers. Under Participants, enter the starting number of this range of ports in the Starting UDP port for audio calls field.
  6. The Lotus Sametime Media Manager listens for inbound video streams from clients on a range of 100 UDP port numbers. Under Participants, enter the starting number of this range of ports in the Starting UDP port for video calls field.
  7. Click OK.
  8. Restart the Lotus Sametime Media Manager.

Managing multiple audio and video streams

The IBM® Lotus® Sametime® Media Manager manages multiple audio and video streams in a meeting.

About this task

The Lotus Sametime Media Manager scans the meeting participants and locates the person currently speaking (transmitting audio packets). The Lotus Sametime Media Manager performs switching operations as different people speak during a meeting. When a meeting participant speaks, the Lotus Sametime Media Manager locks onto that client’s audio stream and distributes that stream to all other clients in the meeting. When a participant stops speaking, the Lotus Sametime Media Manager waits for a brief period of time, and then begins scanning for the other active audio clients.

The video follows the audio. When the Lotus Sametime Media Manager switches to a new audio source (speaking person), the Lotus Sametime Media Manager through its connections to the clients, ensures that the icon indicating the current speaker is properly updated for all clients. After this update, the Lotus Sametime Media Manager sets the video source to the person currently speaking. It is important to ensure that the video does not switch too quickly. Rapid video switching reduces usability. You can control the time interval that must pass before the video switches to the new speaking person.

Note: If the current speaker does not have video capabilities or has the video window paused, the Lotus Sametime Media Manager send’s the next loudest speaker’s video as the active speaker to all participants.

By default, the Lotus Sametime Media Manager can lock onto and broadcast a maximum of five audio streams at the same time. In a meeting, if five people speak at the same time, it is possible for all meeting participants to simultaneously hear five people speaking. The Lotus Sametime Media Manager designates the audio stream that has been transmitting the longest (generally, the person who started speaking first) as the primary audio stream. The source of the primary audio stream is also the source of the video stream. Audio and video services provided by the Lotus Sametime Media Manager have been tested and optimized for sessions with six participants. The actual number of participants will vary based on network and environmental conditions. The higher the number of switched audio streams, then the more bandwidth that is required.

In meetings, especially in large meetings, IBM recommends that participants, who are not talking, mute their speakers to reduce noise.

Procedure

  1. Log in to the Integrated Solutions Console.
  2. Click Sametime System Console > Sametime Servers > Sametime Media Manager.
  3. In the Sametime Media Managers list, click the deployment name of the Lotus Sametime Media Manager.
  4. Click the Configuration tab.
  5. Under Presence, type a number between 2 and 16 in the Number of switched audio streams field to change the number of simultaneous audio streams.

    Note: The more switched audio streams, the more people are heard simultaneously in a meeting. Meetings could become quite noisy, so use caution when you increase this number.

  6. Under Set the time in milliseconds before switching to the next active speaker, select a number in Video switching wait time to control the time interval that must pass before the video switches to the new speaking person.
  7. Click OK.
  8. Restart the Lotus Sametime Media Manager.

Changing the SIP transport protocol in the Sametime Media Manager

You can change the transport protocol that IBM® Lotus® Media Manager uses for the SIP Proxy and Registrar.

About this task

SIP makes use of elements called proxy servers to help route requests to the user’s current location, authenticate and authorize users for media services, and provide Lotus Sametime® Media features to users. SIP also provides a registration function that allows users to send their current locations for use by proxy servers.

The transport protocol determines the network transport mechanism to use for sending SIP messages. The SIP proxy application examines all requests sent by the Lotus Sametime Media Manager to determine whether a given request is sent by an appropriate proxy application. All requests are routed according to the transport protocol defined here.

Procedure

  1. Log in to the Integrated Solutions Console.
  2. Click Sametime System Console > Sametime Servers > Sametime Media Manager.
  3. In the Sametime Media Managers list, click the deployment name of the Lotus Sametime Media Manager.
  4. Click the Configuration tab.
  5. Under Server Integration, select a Transport protocol of UDP or TCP.
  6. Set how frequently in seconds you want SIP to check if a client is still connected. Enter a number between 30 and 300 in the Session expiration field.
  7. Click OK.
  8. Restart the Lotus Sametime Media Manager.

Managing media codecs

You can manage type of media codecs used in meetings on the IBM® Lotus® Sametime® Media Manager.

About this task

A codec compresses streaming data, such as audio or video, on the transmitting side and decompresses it for playback on the receiving side. Codecs reduce the amount of bandwidth required to send streaming data. Generally, higher compression conserves more bandwidth. Higher compression also results in poorer audio or video quality and requires more resources to compress and decompress the data streams.

You can change the type audio and video codecs.

Procedure

  1. Log in to the Integrated Solutions Console.
  2. Click Sametime System Console > Sametime Servers > Sametime Media Manager.
  3. In the Sametime Media Managers list, click the deployment name of the Lotus Sametime Media Manager.
  4. Click the Configuration tab.
  5. Prioritize the audio codecs by using the Up and Down buttons to move the audio codecs in the list.

    Lotus Sametime Media Manager supports the following audio codecs:

    • ISAC – Internet Speech Audio Codec (iSAC) is a wideband and adaptive bit rate codec. The bit rate ranges from 10 to 32 kbps (Kilobit per second) depending on the available network bandwidth. This is the default codec.
    • iLBC – Internet Low Bit-rate Codec (iLBC) is a narrowband low bit rate speech codec. It requires 15.2 kpbs bandwidth.
    • G.722.1 – Popular wideband audio codec that operates at one of three selectable bit rates: 32000, 24000, 16000. G7221 is a licensed royalty-free standard audio codec providing high quality, moderate bit rate audio coding.
    • G.711 – Old and widely supported narrowband codec. It requires 64 kbps bandwidth but consumes less CPU to process.
  6. Prioritize the video codecs by using the Up and Down buttons to move the video codecs in the list.

    Lotus Sametime Media Manager supports two video codecs:

    • H264 – Also known as AVC and MPEG-4 part 10. It provides high quality, block-oriented, motion-compensation-based video codec for video conferencing. It supports the Baseline Profile without Flexible Macroblock Ordering (FMO).
    • H263 – A legacy codec and lower quality than H264.
  7. Click OK.
  8. Restart the Lotus Sametime Media Manager.

About thangletoan

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Posted on 02/06/2012, in Audio, Công nghệ và Giáo dục, Chính sách CNTT, IBM, IBM Lotus Domino, Meeting Center, Sametime, Sống và đam mê khoa học, Video, Web Conferencing and tagged . Bookmark the permalink. Để lại bình luận.

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