sametime 8.5 audio video
Post on 21-Apr-2017
11.828 Views
Preview:
TRANSCRIPT
IBM Sametime 8.5.2
Audio / Video
Vincent Perrin | IBM Certified Collaboration Solutions
Architect
June, 2011
Realtime
Speaker
Agenda
IBM Sametime 8.5.2 Audio / Video Capabilities
IBM Sametime 8.5.2 new Audio / Video Capabilities
Audio Partner Integration
Video Partner Integration
Headset Providers
Sametime Connect Client with Internal Audio Video
Point to point video calls
Multipoint video calls in voice-activated switching mode
Integrated with Meetings (only in Connect Client)
Video codecs: H.264, H.263
Audio codecs: G.722.1, G.711, iSAC, iLBC
Encryptions: RC2, SRTP
RC2 is a block cipher designed by Ron Rivest in 1987 (Rivest Cipher)
Sametime Connect Client with External Bridge
External partners can use Sametime Connect Client
Consistent user interface
Seamless integration
Support multipoint video layouts from external bridge
Sametime Media Manager Overview
SIP Proxy/Registrar
Conference Manager
Packet Switch
Bandwidth Manager *
Sametime ClientsStand-alone client
Embedded client
Meeting client (Embedded in Connect or Web *)
* 8.5.2
Proxy / RegistrarConferenceManagerPacketSwitchBandwidthManagerCommunityServerMeetingServerSametimeClient
VP
SIP
Media
HTTP
VP
SIP Proxy / Registrar
Maintains a registry of all clients with their location, i.e., IP address
Maintains a registry of all conferences with their location
Used to route SIP messages to the proper destination
Proxy / RegistrarConferenceManagerPacketSwitchBandwidthManagerCommunityServerMeetingServerSametimeClient
VP
SIP
Media
HTTP
VP
Conference Manager
Signaling focal point
Manages the sessions
Manages the Packet Switches
Used in 1 to 1 sessions as well N-Way conferences and meetings
Proxy / RegistrarConferenceManagerPacketSwitchBandwidthManagerCommunityServerMeetingServerSametimeClient
VP
SIP
Media
HTTP
VP
Packet Switch
The Media focal point
Used in N-Way conferences and meetings
Controlled by the Conference Manager
Is responsible for distributing the incoming media from conference participants to all other participants
Can be replaced by 3rd party implementations (MCU/Bridge)
Proxy / RegistrarConferenceManagerPacketSwitchBandwidthManagerCommunityServerMeetingServerSametimeClient
VP
SIP
Media
HTTP
VP
Bandwidth Manager
Leverage bandwidth data in SIP SDP
Centralized call rate provisioning with policiesProtects network and constrains bandwidth usage
Defines Class of users and location call rate policy
Logical network topology modelCall admission control based on location in network
A call is either succeeded, failed, or renegotiated
Proxy / RegistrarConferenceManagerPacketSwitchBandwidthManagerCommunityServerMeetingServerSametimeClient
VP
SIP
Media
HTTP
VP
Agenda
IBM Sametime 8.5.2 Audio / Video Capabilities
IBM Sametime 8.5.2 new Audio / Video Capabilities
Audio Partner Integration
Video Partner Integration
Headsets Provider
Sametime 8.5.2 audio / video management
Web Audio / Video
NAT traversal
Bandwidth management
Multiple 3rd party A/V partner integration and management
Sametime Web Audio / Video
1700MXP_cat
V500Cameras
SIPWeb Client Meetings UINative Softphone
HTTP AJAX RequestJSONResponse(Asynchronous)ActiveX/NPPlugin(XML)Window callback(POST message)
Call Notification callbackTelephony REST APIsCall Control Service Plugin
Sametime Proxy Server
Proxy RegistrarConference ManagerPacket SwitcherVideo Engine
Sametime Media Server
Media RTPExtensible and Secure Architecture
SIP
Partner/home Network
Corporate Network
Public Network
DMZ
Sametime 8.5.2 NAT traversal for audio/video
Internet
NAT
Router
New:Sametime TURN server
Sametime clients(rich or web)*
Sametime Media Manager
Sametime clients(rich or web)*
Sametime clients(rich or web)*
* Requires updated Sametime 8.5.x clients (rich or web)
Enables audio and video connectivity across firewalls
Supports ICE/STUN/TURN standards
Firewall / NAT Traversal is a potential road block to deployment
Extranet collaboration (via the DMZ)
Internal separation of divisions
The answer is standards
Control signaling and media must be considered separately
For control signaling, SIP proxies must be strategically located to act as application-layer routers for all endpoints
For A/V media ICE/STUN/TURN standards come to the rescue
New Sametime components in Sametime 8.5.x
ICE/STUN/TURN support built into Sametime 8.5.x A/V clients and servers
New TURN server component (similar function to Sametime Reflector in 8.0.2)
Internal Users Only - No NAT
CommunityServer
MediaServers
MeetingServer
ST ProxyServer
Alice
Bob
Internal Users Only with Firewall / NAT
CommunityServer
MediaServers
MeetingServer
ST ProxyServer
TURNServer
Alice
Bob
NAT
NAT
DMZ Option I
InternalUsers (LAN)
DMZ
ExternalUsers (Internet)
CommunityServer
MediaServers
MeetingServer
ST ProxyServer
TURNServer
Alice
Bob
DMZ Option II
InternalUsers (LAN)
DMZ
ExternalUsers (Internet)
CommunityServer
MediaServers
MeetingServer
ST ProxyServer
TURNServer
SIP ProxyEdge Server
HTTPReverse Proxy
CommunityMux
Alice
Bob
Sametime 8.5.2 Bandwidth Manager
Fernando (Brazil)
Ted VP (US)
Amadou (France)
Gail VP (China)
Protects network by restricting bandwidth used for Sametime audio/video
Manages calls to available bandwidth at each location
Uses bandwidth policies based on classes of users
Note: Final product features and user interface are subject to change
Sametime 8.5.1 and earlier
Leverages existing network tools to identify/manage different media packets types to follow network QoS rules
Sametime AV packets are tagged to allow identification by QoS system
Sametime 8.5.x controls use and protects network bandwidth
New feature of Sametime Media Manager
Call admission control based on location in network
Protects network and constrains bandwidth usage
Call succeeds, fails, or is renegotiated (e.g., video to audio only)
Define Class of users and location call rate policy
Based on defined logical network topology model
Sametime Bandwidth Manager
Bandwidth Manager provides network capacity (bandwidth) control and protection via class-of-user and location-based call rate policies and network topology modeling
The back-office administrator can control access to available bandwidth on each segment of the network topology by provisioning call rate policies to limit the network bandwidth for audio and video data
The policy constrains the amount of bandwidth available for audio and video so that audio and video calls won't interfere with other traffic on the network
Policy is associated with sites or groups of sites in the network topology, specific users or classes of users, or specific predefined groups of users
The administrator can monitor usage of bandwidth in order to tune topology and policy settings
Message Flow (relationship to SIP Proxy)
1. The message is first sent from endpoint A to the SIP Proxy2. The SIP Proxy routes the message to BWM according to the routing rules3. BWM inserts its Contact URL and modifies the SDP payload based on the applied policies then sends the message back to the SIP Proxy4. The Proxy Registrar sends the modified message to endpoint BThe endpoints can be either the Media Manager Conference Focus, Packet Switcher, or any of the supported clients depending on which component initiated the SIP flow
All subsequent messages in the same SIP dialog follow the same route
The SIP Proxy avoids endless looping since the Contact header indicates that BWM server has already been visited
INVITE sip:BFrom: sip: A@domainTo: sip:B@domainContact: sip:
payload SDP
Endpoint A
INVITE sip:BFrom: sip: A@domainTo: sip:B@domainContact: sip:
payload SDP - modified
1
2
3
4
Endpoint B
SIP Proxy
BandwidthManager
How Bandwidth Manager Affects A/V Calls
Based on configured policy at call setup time, Bandwidth Manager decides if there is enough network capacity to complete the call
If not, the Bandwidth Manager might:Remove video media from the call if applicable
Deny the call
Bandwidth manager also can modify certain call parameters based on policy in order to reduce the impact of individual calls on network usage:Reduction of audio quality by removing offered or answered audio codecs
Reduction of video resolution by modification of any supplied b-parm values
Normally users do not notice Bandwidth Manager in a well-tuned network and a proper and accurate topology model
Sametime 8.5.2 multiple A/V partner integration
Note: Final product features and user interface are subject to change
Partner audio bridge connector
Partner video conference connector
Sametime Media Manager
Allows Sametime native + a 3rd party audio + a 3rd party video service
Lets users select appropriate service for each call or conference
Manages access to each service via policies
Voice only
or
Voice+ video
Sametime 8.5 introduced SIP based Media Manager
Enables native audio/video conferences between Sametime clients
3rd party extensions enable interop with audio/ video conference bridges
e.g., Polycom, Radvision, iLink (Tandberg), Avaya, PGI, Avistar, others
Sametime Unified Telephony Lite 8.5.2
Simpler SUT deployment option (licence and pricing terms to follow)
Make / receive video and voice calls from Sametime 8.5.2 a/v client
Limited in-call control
Uses Sametime Media Manage/ SIP Proxy server properly configured with a SIP-trunk-connection (SUT Telephony Application Server and Telephony Control Server are not required)
Upgrade-able to full SUT
At 8.5.2 GA, IBM will certify major SIP communication environments
IP PBX's and SIP based audio and video conference bridges
Other SIP infrastructure configuration certifications will follow after GA
Make / receive voice calls from Sametime 8.5.2 a/v client
Call video endpoints or video MCUs
Call a telephone endpoints or audio conference bridges
Within a call: mute/unmute, raise/lower volume, start/stop video, leave call
Sametime Unified Telephony
Features Compared
All of SUT Lite features plus a richer telephony experience
Single number service
On-a-call presence status
Multiple device support
Contextual Incoming call rules
Transfer calls between devices
Hold, Transfer, merge calls
Visual audio conferencing
Moderator conf controls
Works with multiple PBXs
SUT Lite
SUT
SUT 8.5.2 Lite deployment option
SIP endpoints
SametimePresence / IMServer
Sametime user
SIP Audio/ Video Conference bridge
PSTNGateway
H.323Gateway
H.323 legacy video rooms
3rd Party SIPInfrastructure
External phones
MeetingsServer
TDM
SIP
Sametime Media Manger
Other user
Agenda
IBM Sametime 8.5.2 Audio / Video Capabilities
IBM Sametime 8.5.2 new Audio / Video Capabilities
Audio Partner Integration
Video Partner Integration
Headsets Provider
Cisco Unified Communications
with IBM Lotus
Cisco IP Telephony With IBM
IP Communicator Click to Call with Lotus
Sametime *
Phone Control with Lotus Sametime *
Click to Call & Conference with Lotus Sametime *
Cisco TelePresence & Conferencing with IBM
TelePresence Setup from Lotus Notes
WebEx/Unified MeetingPlace Setup/Attend from Lotus Notes
WebEx/Unified MeetingPlace Click to Conference from Lotus Sametime Instant Messaging *
Cisco Messaging with IBM
Unified Messaging with Lotus Notes
Unified Messaging with Lotus
Sametime *Cisco Presence with IBM
Phone Presence with Lotus Sametime *
Unified Presence Federation with Lotus Sametime
The capabilities in each category are shown here more detail on each is available in following slides
Cisco Unified Communications
with IBM Lotus Sametime
NOTE: all client plug-ins shown here
This graphic shows the user experience of a number of the Sametime client integrations IP Communicator Click to Call with Lotus SametimeCisco Phone Control with Lotus SametimeCisco Phone Presence with Lotus SametimeUnified Messaging with Lotus SametimeWebEx Click to Conference with Lotus Sametime Instant Messaging
Alcatel / Lotus Notes/Domino
My Instant Communicator for Lotus Sametime: Phone presence, notifications, call control, visual voice mail,
Avaya/IBM Integrations
multi-vendor best-of-breed solutions
INTEGRATIONS
Click-to-Communicate
Federated Presence
Unified messaging
Unified conferencing
Mobility
Avaya ACEAvaya FlareApplication Enablement ServicesAvaya Aura Communication ManagerCS 1000Avaya Aura Presence ServicesModular MessagingMeeting ExchangeAvaya Aura ConferencingAvaya one-X CommunicatorAvaya one-X Speech
Sametime ServerSametime Connect ClientDominoNotesSametime Web ConferencingLive NamesConnectionsQuickrWebSphere - Lombardi
SameTime
Server
Web Services
Virtual Places (VP) Protocol
ACE
Server
(Standalone or High Availability)
Application Integration Engine (AIE)
Server
Virtual Places (VP) Protocol
CS1000
CISCO CUCM
Avaya
AES/CM
TR87 & SIP
JTAPI & SIP
TR87 & SIP
Tandberg
VCS
SIP
Media Application Server (MAS)
(NMC)
TCSPI
Avaya ACE and IBM Integration
ACE Server, ACE TCSPI and AIE to support IBM Desktop
Agenda
IBM Sametime 8.5.2 Audio / Video Capabilities
IBM Sametime 8.5.2 new Audio / Video Capabilities
Audio Partner Integration
Video Partner Integration
Headsets Provider
What is the Polycom-IBM Integration?
End-users maintain familiar Lotus Sametime interfaces to enable voice and video collaboration
Click-to-conference from Lotus Sametime Connect client or from web meeting
Integrated with standards-based H.323(and H.320, PSTN) endpoints
Multipoint continuous presence (versus active speaker)
Polycom RMX
Integration Components
IBM/Polycom Sametime Architecture
ScaleResources and Redundancy
RMX 2000
RMX 4000
RMX 1500
ResolutionResources
Voice1440
CIF/HD VSW360
SD/4CIF240
HD 720p120
HD 720p 60fps60
HD 1080p60
ResolutionResources
Voice720
CIF/HD VSW180
SD/4CIF120
HD 720p60
HD 720p 60fps30
HD 1080p30
ResolutionResources
Voice360
CIF/HD VSW90
SD/4CIF60
HD 720p30
HD 720p 60fps15
HD 1080p15
~1800 Sametime Users*
~3600 Sametime Users*
~7200 Sametime Users*
*Assume 256k or 384kCIF desktop video. Assumes one concurrent Lotus Sametime video client call for every 10 users. Assume half of these calls would be on a bridge. This will need to be adjusted due to Lotus Sametime architecture, time zones diversity and other. factors
Conference Infrastructure Scale
As a rule of thumbs, for calculating bandwidth, assume one concurrent lotus sametime video client call for every 10 users. Assume half of these calls would be on a bridged call.
This is the current snapshot of Polycom conference platform.Polycom 1500 is a closed box so there is no hardware expansion. For RMX 2000 and RMX 4000, you can add media modules to increase systems capacity.For typical deployment, Lotus Sametime users (computer) will be connected at CIF resolution. (Louts Sametime supports up to 1080P high definition, but it requires both powerful CPU and bandwidth. Most typical deployment for hundreds or thousands of computer video users is CIF-equivalent. (352 x 288)
Polycom RMX can manage resources dynamically, which is to say, consume only the resources required to support the resolution used. Therefore, if there are more computer video users than room video conference system with HD video, then the resources are optimized so that more computers can be connected simultaneously.
Matt
John
Chris
Peter
IM with multiple people
Lets do video call.Matt start a video call
Sametime creates a conference on RMX
Matts ST client places SIP call to RMX
Steve: Accept the call from ST client
Chris: Accept the call from ST clientPeter: Accept call from HDX in the pull-down listRMX calls out to HDX (h.323, etc) and ST clients (SIP) dial in
Seamless Escalation from IM to Multiparty Video
Use Case 1: Click-to-conference
Select contacts and 1) start chats and escalate to video call 2) directly start a video call
Lotus Sametime users receive incoming call notification and select device from which to receive the call. Users can simply add
Polycom RMX hosts the voice and video multiparty calls with continuous screen for rich conference experience
Use Case 2: Web Collaboration
Enter a Sametime meeting room (ad hoc or scheduled).Invite others (drag & drop contact names into Meeting Room or choose contacts from Meeting Room UI)
The recipient of the meeting joins the meeting room.
Participants dial into the Polycom RMX from within the meeting room window
logoRV.pngRadvision Solution Architecture
Supported Use Cases
User Scenarios
ST 8.5 Infrastructure
SCOPIA Infrastructure
P2P Call ST to ST
P2P Call ST to VC
P2P Call ST to Phone/Cell
Multipoint Calls
Escalation from P2P to Multipoint
D:\Projects\ICONS\PNG\VistaICO_Toolbar_Icons\Symbol-Check.pngD:\Projects\ICONS\PNG\VistaICO_Toolbar_Icons\Symbol-Delete.pngD:\Projects\ICONS\PNG\VistaICO_Toolbar_Icons\Symbol-Check.pngD:\Projects\ICONS\PNG\VistaICO_Toolbar_Icons\Symbol-Delete.pngD:\Projects\ICONS\PNG\VistaICO_Toolbar_Icons\Symbol-Check.pngD:\Projects\ICONS\PNG\VistaICO_Toolbar_Icons\Symbol-Check.pngD:\Projects\ICONS\PNG\VistaICO_Toolbar_Icons\Symbol-Delete.pngD:\Projects\ICONS\PNG\VistaICO_Toolbar_Icons\Symbol-Delete.pngD:\Projects\ICONS\PNG\VistaICO_Toolbar_Icons\Symbol-Check.pngD:\Projects\ICONS\PNG\VistaICO_Toolbar_Icons\Symbol-Check.png\\172.16.91.101\public\System 7 launch materials\Resources\iVIEW_V7_Sales_Presentation\pointer.pnglogoRV.png
Complete Sametime Application Integration
Meeting RoomsAd-hoc Meetings
logoRV.png
IBM Lotus Sametime
Integration for Conference Initiation & Presence
Single click conference initiation consistent with other modalities
Automatic presence updates
Automatic guest link via IM for non-Vidyo users
Resides on clients machine, no server component required
Tandberg + Ilink
Ilink is delivering Tandberg Adaptor for ST 8.5.x
This PlugIn is providing Tandberg Video Capabilities Integration into Sametime Connect and Meeting Room.
Feature Set
ClicktoCall ClicktoConference Adhoc Conferencing Meet Me Conferencing Active Conferencing Control Dial In Dial Out Hangup Mute User Mute Conference
IBM/Ilink/ Tandberg Architecture
Agenda
IBM Sametime 8.5.2 Audio / Video Capabilities
IBM Sametime 8.5.2 new Audio / Video Capabilities
Audio Partner Integration
Video Partner Integration
Headsets Provider
Plantronics Plug-In for IBM Sametime
The latest Hands free headset call control between IBM Sametime and Plantronics UC audio devices, provide users with exceptional online meeting experiences with enhanced audio quality and connectivity including:
1. Control Sametime calls remotely from the Headset: Answer/end, mute and volume control features allows you to gain hands-free mobility directly from the audio device if you need to roam away from your desk, or are multi tasking whilst on Sametime /SUT Calls.
2. Put Sametime calls on hold remotely from the headset : Put your Sametime Caller on hold so you can conduct business privately and know your caller is secure and your alternative conversation is confidential
3. Switch between Sametime calls remotely from the headset: Answer other inbound calls including Skype calls whilst multi tasking away from the computer. Dont miss out on connecting with colleagues and customers needing to speak with you simultaneously.
4. Smart Sensor Wearing State: Answers call automatically as soon as headset is (donned) or placed on the ear without the need to press answer button on headset -or on the screen.
5. Mobile telephony presence enhances both Sametime and SUT by eliminating the blind spot in SUT by changing Sametime presence status to On Phone when a user is on a mobile telephone / cell call. This requires no user intervention as the headset solution automatically detects the mobile phone call and changes the presence status in Sametime
Jabra
Jabra, an IBM Business Partner, seamlessly integrates unmatched endpoint audio quality with IBM Sametime and Sametime Unified Telephony. Jabra headsets deliver an enhanced user experience, enabling users to fully control Sametime softphones from the headset while providing hands-free freedom and mobility.
Exceptional sound quality using the latest advances in audio technology combines with ergonomic design to promote productivity and user satisfaction.
Sennheiser
The Sennheiser Call Control plug-in surfaces key functionality within the Sametime user experience by providing a seamless integration of the Sennheiser headset with IBM Sametime.
You benefit from a fully tested solution making use of IBM Sametime together with Sennheiser headsets. All call-control features can be used via the Sennheiser headset, i.e. answering and ending calls, adjusting volume or muting the call. The DW Series wireless headsets provide you with maximum mobility to roam around the office while still being able to communicate.
cropVincent PerrinLotus Collaboration Solutions ArchitectIBM Software Group
17, avenue de l'europeBois ColombesTel +33 677 02 03 54vincent.perrin@fr.ibm.com
Standard Protocols - Media
STUN
Session Traversal Utilities for NAT
An IETF protocol RFC 5389
TURN
Traversal Using Relays around NAT
An IETF extension to STUN RFC 5766
ICE
Interactive Connectivity Establishment
An IETF standard RFC 5245
A procedure used by media end point to establish a valid connection
Uses STUN and TURN
Works through almost all types of NAT and Firewalls
Works in very complicated and challenging networks
Finds the shortest / most efficient path available
2010 IBM Corporation
2011 IBM Corporation
Polycom 2010
Polycom 2010
Click to edit Master title style
Click to edit Master text styles
Second level
Third level
Fourth level
Fifth level
Polycom 2010
Polycom 2010
Click to edit Master title style
logoRV_samall.png
Title- Limited to 46 Characters/Line Can Also Extend to a Second Line When Necessary
Click to edit Master text styles
Second level
Third level
21/07/2011
This presentation contains materials that are either copyright 2008 IBM Corporation or copyright 2008 Cisco Systems, Inc. All rights reserved.
Click to edit Master title style
Click to edit Master text styles
Second level
Third levelFourth levelFifth level
2010 IBM Corporation
Click to edit the outline text formatSecond Outline LevelThird Outline LevelFourth Outline LevelFifth Outline LevelSixth Outline LevelSeventh Outline LevelEighth Outline LevelNinth Outline Level
IBM Software
2011 IBM Corporation
2011 IBM Corporation
2010 IBM Corporation
top related