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  • RC2 is a block cipher designed by Ron Rivest in 1987 (Rivest Cipher)
  • 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)
  • 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 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
  • The capabilities in each category are shown here … more detail on each is available in following slides
  • This graphic shows the user experience of a number of the Sametime client integrations … IP Communicator Click to Call with Lotus Sametime Cisco Phone Control with Lotus Sametime Cisco Phone Presence with Lotus Sametime Unified Messaging with Lotus Sametime WebEx Click to Conference with Lotus Sametime Instant Messaging
  • 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 system’s 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.

Sametime 8.5 Audio Video Sametime 8.5 Audio Video Presentation 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
  • 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 Clients
      • Stand-alone client
      • Embedded client
      • Meeting client (Embedded in Connect or Web *)
    * 8.5.2 Proxy / Registrar Conference Manager Packet Switch Bandwidth Manager
  • 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 / Registrar Conference Manager Packet Switch Bandwidth Manager
  • 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 / Registrar Conference Manager Packet Switch Bandwidth Manager
  • 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 3 rd party implementations (MCU/Bridge)
    Proxy / Registrar Conference Manager Packet Switch Bandwidth Manager
  • Bandwidth Manager
    • Leverage bandwidth data in SIP SDP
    • Centralized call rate provisioning with policies
      • Protects network and constrains bandwidth usage
      • Defines “Class of users” and location call rate policy
    • Logical network topology model
      • Call admission control based on location in network
      • A call is either succeeded, failed, or renegotiated
    Proxy / Registrar Conference Manager Packet Switch Bandwidth Manager
  • 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
  • Extensible and Secure Architecture SIP Web Client Meetings UI Native Softphone HTTP AJAX Request JSON Response (Asynchronous) ActiveX/NPPlugin (XML) Window callback (POST message) Call Notification callback Telephony REST APIs Call Control Service Plugin Sametime Proxy Server Proxy Registrar Conference Manager Packet Switcher Video Engine Sametime Media Server Media RTP SIP 1700MXP_cat V500 Cameras
    • Sametime 8.5.2 NAT traversal for audio/video
      Partner/home Network
      Corporate Network
      Public Network
      DMZ
      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 a udio and video connectivity across firewalls
    • Supports ICE/STUN/TURN standards
  • Internal Users Only - No NAT Alice Bob Community Server Media Servers Meeting Server ST Proxy Server
  • Internal Users Only – with Firewall / NAT Alice Bob Community Server Media Servers Meeting Server ST Proxy Server TURN Server NAT NAT
  • DMZ – Option I Internal Users (LAN) DMZ External Users (Internet) Alice Bob Community Server Media Servers Meeting Server ST Proxy Server TURN Server
  • DMZ – Option II Internal Users (LAN) DMZ External Users (Internet) Alice Bob Community Server Media Servers Meeting Server ST Proxy Server TURN Server SIP Proxy Edge Server HTTP Reverse Proxy Community Mux
    • 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 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 Proxy 2. The SIP Proxy routes the message to BWM according to the routing rules 3. BWM inserts its Contact URL and modifies the SDP payload based on the applied policies then sends the message back to the SIP Proxy 4. The Proxy Registrar sends the modified message to endpoint B
    • The 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:B From: sip: [email_address] To: sip: [email_address] Contact: sip:<IP A> payload SDP Endpoint A INVITE sip:B From: sip: [email_address] To: sip: [email_address] Contact: sip:<IP BWM> payload SDP - modified 1 2 3 4 Endpoint B SIP Proxy Bandwidth Manager
  • 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
    • 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 Media Manager
    • 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
      Sametime Presence / IM Server
      Sametime user
      SIP Audio/ Video Conference bridge
      PSTN Gateway
      H.323 Gateway
      H.323 legacy video rooms
      3 rd Party SIP Infrastructure
      External phones
      Meetings Server
      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
  • Cisco Unified Communications with IBM Lotus Sametime NOTE: all client plug-ins shown here
  • 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 Avaya ACE Avaya Flare Application Enablement Services Avaya Aura Communication Manager CS 1000 Avaya Aura Presence Services Modular Messaging Meeting Exchange Avaya Aura Conferencing Avaya one-X Communicator Avaya one-X Speech Sametime Server Sametime Connect Client Domino Notes Sametime Web Conferencing Live Names Connections Quickr WebSphere - Lombardi INTEGRATIONS Click-to-Communicate Federated Presence Unified messaging Unified conferencing Mobility
  • SameTime Server Web Services Virtual Places (VP) Protocol ACE Server (Standalone or High Availability) Application Integration Engine (AIE) Server Virtual Places (VP) Protocol TR87 & SIP JTAPI & SIP TR87 & SIP SIP Media Application Server (MAS) (NMC) TCSPI Avaya ACE and IBM Integration ACE Server, ACE TCSPI and AIE to support IBM Desktop CS1000 CISCO CUCM Avaya AES/CM Tandberg VCS
  • 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
  • RMX 2000 RMX 4000 RMX 1500 Resolution Resources Voice 1440 CIF/HD VSW 360 SD/4CIF 240 HD 720p 120 HD 720p 60fps 60 HD 1080p 60 Resolution Resources Voice 720 CIF/HD VSW 180 SD/4CIF 120 HD 720p 60 HD 720p 60fps 30 HD 1080p 30 Resolution Resources Voice 360 CIF/HD VSW 90 SD/4CIF 60 HD 720p 30 HD 720p 60fps 15 HD 1080p 15 ~ 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 Scale Resources and Redundancy
  • Matt John Chris Peter
    • IM with multiple people
    “ Let’s do video call.”
    • Matt start a video call
    • Sametime creates a conference on RMX
    • Matt’s ST client places SIP call to RMX
    • Steve: Accept the call from ST client
    Chris: Accept the call from ST client Peter: Accept call from HDX in the pull-down list
    • RMX 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.png Radvision Solution Architecture
    • 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:ProjectsICONSPNGVistaICO_Toolbar_IconsSymbol-Check.png D:ProjectsICONSPNGVistaICO_Toolbar_IconsSymbol-Delete.png D:ProjectsICONSPNGVistaICO_Toolbar_IconsSymbol-Check.png D:ProjectsICONSPNGVistaICO_Toolbar_IconsSymbol-Delete.png D:ProjectsICONSPNGVistaICO_Toolbar_IconsSymbol-Check.png D:ProjectsICONSPNGVistaICO_Toolbar_IconsSymbol-Check.png D:ProjectsICONSPNGVistaICO_Toolbar_IconsSymbol-Delete.png D:ProjectsICONSPNGVistaICO_Toolbar_IconsSymbol-Delete.png D:ProjectsICONSPNGVistaICO_Toolbar_IconsSymbol-Check.png D:ProjectsICONSPNGVistaICO_Toolbar_IconsSymbol-Check.png 172.16.91.101publicSystem 7 launch materialsResourcesiVIEW_V7_Sales_Presentationpointer.png logoRV.png
      Supported Use Cases
  • Complete Sametime Application Integration logoRV.png Meeting Rooms Ad-hoc Meetings
  • 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 client’s machine, no server component required
  • Tandberg + Ilink
    • Ilink is delivering Tandberg Adaptor for ST 8.5.x
    • This Plug‐In is providing Tandberg Video Capabilities Integration into Sametime Connect and Meeting Room.
    • Feature Set
      • • Click‐to‐Call
      • • Click‐to‐Conference
      • • Ad‐hoc 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 late st 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. Don’t 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.
  • crop Vincent Perrin Lotus Collaboration Solutions Architect IBM Software Group 17, avenue de l'europe Bois Colombes Tel +33 677 02 03 54 [email_address]
  • 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