sametime 8.5 audio video

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


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


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)


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


VP

SIP

Media

HTTP

VP

Agenda





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



* 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





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





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*



*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

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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





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 [email protected]

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


Polycom 2010

Polycom 2010

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Polycom 2010


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21/07/2011

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