Keep Calm and Call On! IBM Sametime Communicate Softphone Made Simple. Frank Altenburg, IBM

Keep Calm and Call On! IBM Sametime Communicate Softphone Made Simple Frank Altenburg, IBM

Agenda Voice and Video an effective way to do business! Sametime Softphone Computer is your phone! Sametime Voice and Video options Sametime Softphone your real-time social communication tool Sametime Softphone usage options based on your business requirements

Voice and Video an effective way to do business! Integrate real-time social communications into your business environment Instantly get connected to the person on the other end- with just one click from your Sametime client Helps to meet the ongoing demands of your everyday business Provides a unified AV user experience

What is a softphone? A component of the Sametime Connect client Acts as an interface allowing you to dial numbers and carry out other phone functions It is an integrated Sametime Computer Phone - supports computer-to-computer, computer-to-telephone, or telephone-to-computer calls

Communicate through softphone - Computer is your phone!

Communicate through softphone... Allows for switching from voice to video and back with customizable layout option Provides high definition audio and video output Generates continuous audio and video presence* Experience high definition audio and video* - Software Multipoint Control Unit (MCU) enables up to six people in your video chat, simultaneously - Industry-leading H.264 Scalable Video Coding (SVC) streams video efficiently through varied bandwidths - Video Manager manages a pool of MCUs Point-to-point audio and video on mobile devices Continuous-presence audio and video for mobile devices (* Only available in Sametime 9.0 and above)

Sametime 9.x Voice and Video options Sametime Communicate - for one-to-one video and voice chat Sametime Conference - for continuous presence, multi-way video chat

Sametime Voice and Video options Voice and video for chat Voice and video for meetings Voice and video in the extranet for anonymous access Voice and video in the extranet for employees without VPN Voice and video systems from vendors Voice and video with NAT Click-to-conference using a third-party audio bridge Voice and video with direct dialing Geographically distributed Video MCU Servers

Deployment scenario example #1

Deployment scenario example #2

Deployment scenario example #3

Sametime AV details The standards-based (STUN, TURN, ICE) NAT traversal enables AV usage to transcend an organization's firewalls Bandwidth Manager makes rich media collaboration feasible by helping maintain network QoS for critical applications. Smartphone users have more options now Support for dual adapters into a single Sametime infrastructure - improved flexibility for integrating audio conferencing and enterprise video. Virtualization of IBM Sametime Unified Telephony for both the TAS and TCS servers allows you to take advantage of consolidated IT infrastructure and hosted environments - helps reduce operating costs, increases resource utilization

Protocols used between AV components LDAP and Media Manager: TCP Community Server and Media Manager: TCP SIP PR, Conference Manager, Video Manager and Bandwidth Manager: SIP Internal client and Community Server: VP Internal client and Meeting Server: TCP Internal client and Video Manager and SIP Proxy Registrar: UDP

Sametime 9 AV deployment components

Sametime Audio/Video Components Peer-to-Peer Audio/Video (Sametime Communicate)

Sametime Audio/Video Call Flow Peer-to-Peer Audio/Video (Sametime Communicate)

Sametime Audio/Video Components Multipoint Audio/Video (Sametime Complete)

Sametime Audio/Video Call Flow Multipoint Audio/Video (Sametime Complete)

Each VMGR LB routes to other VMGRs based on load Client CF + PR VMGR LB/sprayer name updated in stavconfig.xml. This is updated from Admin pages/ssc GA. Plan creation order matters I P S P R A Y E R MCU farm VMGR V M G R L B MCU farm VMGR VM GR LB VMGR VM GR LB VMGR figures out Which MCU to route to MCU farm

Different types of softphone calls Standard Sametime AV SIP-based calling (formerly called Sametime Unified Telephony Lite) Sametime Unified Telephony Sametime AV calls using external third-party adapters

Sametime Standard AV - How calling with Softphone works Uses either peer-to-peer or multi-way conversations In peer-to-peer, at login, a user's information goes to the SIP Proxy/Registrar SIP PR is like a digital PBX that responds to things like who's available, who can I call, and who's calling me Users can send a SIP invite request to SIP Proxy/Registrar Users send information via the Session Description Protocol (SDP) about the IP and port numbers and what type of call they want to have The call begins with the media stream flowing peer-to-peer through IPs and ports negotiated Conference calls with more than two people - all the media traffic goes through the Video MCU

SIP-based calling (formerly SUT Lite) Provides Sametime Unified Telephony experience using your existing Session Initiation Protocol (SIP) infrastructure Turn the Sametime Connect client into a standards-based SIP softphone Users can place and receive calls to external phone numbers, video conferencing systems or video conferencing endpoints from their desktop. You can use VoIP, eliminating the need for desk phones, long distance charges Configuration of the SIP trunking feature with third party audio/video equipment Note: The SIP trunk is a two way path between Sametime Media Manager and the video or audio application

Things to know about SIP-based calling... Allows users to make and receive SIP-based phone calls from their installed Sametime clients with a third party audio/video bridge Does not include incoming call routing, device hand-off, and so on It relies on the Media Manager No new servers required Configure SIP trunks from the SIP Proxy/Registrar component Separately licensed feature - builds on your Sametime Standard deployment to add SIP-based telephony features

Deploying SIP-based calling using softphone Allows Sametime users to make SIP-based phone calls to phone numbers or video conferencing end points from their installed Sametime clients. Deploying SIP-based calling requires: - Enabling the feature - Configuring Sametime to route outgoing calls to the appropriate SIP endpoints - Configuring each SIP endpoint to route incoming calls to Sametime (You need to configure the SIP trunks from the SIP PR to external systems)

SIP-based Calling - Components required Media Manager Conference Manager SIP Proxy/Registrar No TCSPI adapter needed

SIP-based calling deployment

IBM Sametime Unified Telephony (SUT) Softphone Unifies communications and collaboration capabilities without replacing your PBX system Provides the integration of telephony across multi-vendor PBX systems resulting in a unified end-user experience, including integrated softphones; phone and IM presence awareness; call control and rulesbased call management Provides users with a consistent communications experience, with a single unified phone number that can reach any device; telephony presence; incoming call management; click-to-call, click-to-conference; and integrated softphone functions Provides a middleware layer to enable telephony integration with multiple PBX backends to support your existing infrastructure

More on using SUT softphone... Integrates telephony into your company's greater communications platform Telephony integration allows person-to-person as well as multi-person interaction. SUT offers the immediacy of IM with telephone capabilities on user desktops - users can collaborate with one another more effectively. The middleware layer of SUT provides connectivity to multiple telephone systems both IP PBX and legacy TDM systems

Simple SUT deployment

How do external TCSPI adapters work? Sametime supports the use of external TCSPI adapters Sametime Media Manager supports a single internal TCSPI adapter called "Sametime Audio\Video Conferencing" In addition, you can deploy external TCSPI adapters; for example, one to support audio, and another for video Telephony Conferencing Service Provider Interface (TCSPI) is a protocol used to establish audio and video calls.

Sample MyAV adapter integrated with Sametime

Sametime Softphone helps cut telephony costs Dual audio/video interface to deliver both built-in and third-party media capabilities in voice and video chats or in Sametime meetings Conferencing easily integrated with existing hardware leverage your existing infrastructure Industry-standard audio and video codecs provide high-quality built-in AV services for a collaborative experience. Audio/video based on the industry standard Session Initiation Protocol (SIP), improving interoperability with third-party audio/video conferencing systems NAT traversal allows AV to pass through firewalls and NAT'ed environments. This support lets people engage in rich communications with any external parties in their network Note: You can make use of Sametime Bandwidth Manager to allocate AV bandwidth limit for the calls to protect your network for mission critical applications/groups of users

Sametime Clients used in AV Connect client Embedded Notes client Rich meeting client or web meeting client Mobile client

Making calls with SIP-based calling option Difference between making calls with conventional Sametime audio/video capabilities and Sametime Unified Telephony Lite Client is the use of SIP URIs Just like a phone number identifies a telephone user, a SIP URI identifies a computer phone Users can initiate calls from several areas of the UI - enter a phone number or a SIPURI, right-click options, Quickfind, Action Bar icons, Phonebook, Dialing keypad, Call History

Making calls with Sametime Unified Telephony option Make and receive calls using the Sametime softphone or your traditional telephones Call people inside or outside of your organization, even those who do not have IBM Sametime Unified Telephony Get notified whenever someone is calling you Route your calls to other devices, depending on your location or other criteria

Demo

Connect to your social business - Mobile A/V support Click to call Video (1 to 1 only) Click to call Audio (1 to 1 only) VoIP calls using SIP-based calling aka SUT-Lite (outgoing only) SUT calls using VoIP (outgoing only)

Mobile SIP-based calling Softphone Place a call via the Sametime softphone using SIP-based calling to any phone number On phone device, there is no built in keypad in Sametime - user uses the native dialer If a phone number is dialed, and Sametime is logged into a SIP-based calling enabled server, the user will be prompted if they want to use it If the user chooses to make the call via SUT, then the call is placed via SUT or SUT-Lite (whichever is enabled) User can also select device Phone itself as the method of placing calls in the Audio/Video settings

Mobile SUT Calls Using Softphone on mobile devices Previously SUT calls could only be made using a phone number as a ringback In ST 9, we allow SUT calls to connect via the softphone SUT call can occur over the network - similar to the Computer Phone on desktop client SUT uses softphone by default, but can be changed to use another ringback number If a phone number is dialed, and Sametime is logged into a SUT enabled server, the user will be prompted if they want to use SUT

SUT mobile screenshot

Additional Mobile Client Features New softphone capability on ios and Android devices - SUT Calls can now be placed without using cellular minutes - Incoming calls still not supported due to device power management issues

Audio/Video functionality available for Mobile... Audio / Video client support ios 6 and 7 Android 4.0 and greater Audio only on phone devices (iphones and Android phones) video supported for tablets Note: Above features are for ST 9

Concluding remarks... Next time you pick up your phone to make a call, think about what Sametime softphone can do for you!

Questions? 49