Sut852 Func Spec

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Sametime

Version 8.5.2

Sametime Unified Telephony

Functional Specification



ContentsIntroduction..............................................................................................................................................................3

System capabilities summary...............................................................................................................................3

Summary of User Features...................................................................................................................................4

Summary of Administrative Features...................................................................................................................5

Supported Languages...........................................................................................................................................5

Sametime Unified Telephony Architecture..........................................................................................................5

Sametime Unified Telephony Components..........................................................................................................6

Software Licensing.................................................................................................................................... ..........6

Software Integration.............................................................................................................................................6

Sametime Unified Telephony embedded softphone........................................................................................... ......7

Telephony Application Server (TAS).......................................................................................................................8

Telephony Application Server Specifications......................................................................................................8

Media Server..........................................................................................................................................................10

Media Server Conferencing capability...............................................................................................................10

Telephony Control Server (TCS)............................................................................................................ ...............12

Telephony Control Server Specifications...........................................................................................................12

Storage Area Network........................................................................................................................................ ....14

Storage Area Network Specifications..................................................................................................... ...........14

Network Storage Specifications.........................................................................................................14

IP Network Connections....................................................................................................................15

Interoperability Testing Information............................................................................................................... .......17 Notices............................................................................................................................................ ..... ..... ..... ..... ...18

Copyright license...............................................................................................................................................19

Trademarks................................................................................................................................................... .....20

IBM Sametime Unified Telephony 8.5.2 Functional Specification 2



Introduction

IBM® Sametime® Unified Telephony (SUT) software is middleware that integrates a market- leadingSametime unified collaboration solution with a telephony solution across multi-vendor Private BranchExchange (PBX) systems and provides a unified end user experience from either inside the Sametime or Lotus Notes client. The experience includes integrated softphones; phone and IM presence awareness;

call control and rules-based call management across multiple communications systems. By leveragingand extending the existing telephony solutions and unifying them with real-time collaboration tools, anorganization can help users find and reach each other more easily and collaborate working together smarter to speed decision-making and business processes.

System capabilities summary

Features

• A simple, consistent user communications experience on the desktop client

• Sametime and phone presence awareness

• Unified business number

• Rules and alert-based call management• Embedded softphone for all telephony environments

• Initiate a call from Sametime or any application via click-to-call/click-to-conference

• Initiate and manage ad hoc audio conferences

• Platform for unified communications and collaboration

• Works with your existing PBX environment (independent of vendors or PBX types)

Benefits

• Makes users more responsive and efficient by allowing them to access telephony functionalityfrom wherever they are—IM, email, Web pages, and corporate applications.

•

Increases user productivity and lowers training and support costs by providing users a simple,consistent user communications experience including telephony presence, incoming callmanagement and call control, click to call and softphones.

• Helps optimize the value from existing enterprise applications and telephony systems -independent of the enterprise's telephony infrastructure or migration to IP (Internet Protocol) basedtelephony. The software integrates both IP-based and legacy-based PBXes.

• Helps reduce calling costs with softphone, call management and aggregated presence awareness.Calls made through the softphone avoid telephony charges. Call management capabilities candirect calls to a user's preferred device so that colleagues do not have to call a variety of devices tofind the user.




Summary of User Features

Each of the following capabilities offers the users access to telephony functionality:

• Telephony presence – At a glance, users can now see the telephone status of their contacts, alongwith online presence status (Available, Away, In a meeting, Do not disturb, and so forth), making iteasy to know whether it is appropriate to initiate a phone conversation. Users can see if the personthey want to contact is on the phone or off the phone.

• Unified business number – Users have a single unified phone number, but the Sametime UnifiedTelephony allows calls to be routed to virtually any phone device in almost any location.

• Incoming call management – Users can set rules and preferences for where their calls will go (for example, redirect to a mobile phone). Because Sametime software provides inherent presence andlocation awareness, SUT software can automatically set the preferred device based on the user'savailability and location status.

• Incoming call notification – Sametime Unified Telephony notifies the user through Sametime clientwhenever there is an incoming call, so that the user has the opportunity to take the call, deflect it toanother device, or let the call go to voicemail.

• Call control - Users are able to manage their phone calls through the Sametime Unified Telephony'svisual call control interface regardless whether they use softphone, desk or mobile phone. Users cansee the call and connection status, callers business card, and have call controls that enable pausingand resuming the call or transferring the call to another phone device. Users can also forward the callto another person or to terminate the call. The call control interface also allows initiating other meansof collaboration, such as text chat, video session or Web collaboration session.

• Click-to-call and Click-to-conference – Starting a call is intuitive; people can simply select namesfrom the Sametime contacts list or QuickFind, search the contact from the enterprise directory. Usersmay initiate calls or conferences directly from an instant message within Sametime, Notes, or Microsoft® Outlook or Microsoft Office products. Users may also enter any phone number number manually using the pop-up keypad.

• QuickFind – Simply type a name or number and Sametime automatically does a type-ahead search of the enterprise directory.

• Visual ad hoc audio conferencing – Through the graphical call control interface, users can see whois on the call, mute themselves, put themselves on hold and transfer the call to another phone device.Moderators can mute individual participants or mute all, lock the call, remove individuals, and soforth.

• Softphone – Users can use their workstation's built-in audio capability or attached headsets to initiateand receive phone calls.

• Personal phone book – Users can store, view, and manage their personal, non-enterprise directory

listed phone contacts to their personal phonebook. Any contact added into the personal phonebook will turn up in Quick find results.

• One-click conference calling – Users can store their frequently-used conference call numbers toenable one-click conference calling, including conference call passwords.

• Personal call history – All Sametime connected calls are logged into a user's personal call historylog. Calls from the log can be added to the user's phonebook, re-dialed or added to the user's contactlist.




Summary of Administrative Features

• Add or remove users or subscribers to Sametime Unified Telephony.

• Set up secure communication for Sametime Unified Telephony communities.

• Monitor call activity and general server status.

• Set up SIP proxy and registrar properties.

• Manage Sametime Unified Telephony components.

• Start, stop, and update servers.

• Back up servers.

• View the most recent calls on the system for diagnostic purposes.

• Obtaining details on all calls, including duration, time and phone numbers.

• Configure options for joining conferences.

Supported Languages

There are three different areas of the Sametime Unified Telephony which require language support. Eacharea has a different set of languages that is supports:

• The Sametime Unified Telephony servers administrators screens (TAS/TCS administration webapplication screens) support English and German.

• Conferences contain the audio messages used during the conference life cycle for the conferencemanagement: American English, British English, Chinese, Brazilian, French, Italian, Japanese,Korean, and Spanish

• Announcements are the audio messages used during a one-to-one call for the life cycle of the callmanagement: Argentines, Bulgarian, Czech, Danish, German, German Austria, Greek, EnglishAustralia, English India, English South Africa, English USA, English, Spanish Argentina, Spanish

Chile, Spanish Mexico, Spanish, Ethiopian, Finnish, French Canada, French, Croatian, Hungarian,Italian, Latvian, Dutch, Dutch Belgium, Norwegian, Polish, Portuguese. Portuguese Brazil,Romanian, Russian, Slovakian, Sierra Leone, Suriname, Swedish, Turkish, and Chinese.

Sametime Unified Telephony Architecture

Sametime Unified Telephony is a middleware that uses open standards and interfaces to connect withtelephony infrastructure to provide telephony services and events to the Sametime users. It is designed to provide a highly flexible, scalable, and reliable system that can integrate with nearly any telephonysystem.

The Sametime Unified Telephony solution consists of:

• Sametime Unified Telephony software with its embedded softphone

• IBM System x™ (xSeries) servers

o Telephony Control Servers (TCS)

o Telephony Application Servers (TAS)

o Warm standby servers (for redundancy)

o Optional Media Servers (for improved performance)




• Storage Area Network and external SAN storage

• IP network connections:

o between Sametime Unified Telephony servers

o to Sametime server and enterprise directory

o to PBXes and Gateways (SIP trunks)

o to Sametime Unified Telephony softphones

• Interoperability-tested PBXes

• Optional interoperability tested SIP gateways used for PSTN and PBX connections

Sametime Unified Telephony Components

Sametime Unified Telephony 8.5.2 software contains two components:

• IBM Sametime Unified Telephony Call - provides Telephony Application Server (TAS) andMedia Server (MS) software. Sametime Unified Telephony Call includes embedded softphones.

• IBM Sametime Unified Telephony Connect - provides Telephony Control Server software.

Software Licensing

The software licensing model is per user licensing. Client pays per user charge.

IBM Sametime Unified Telephony software license includes a 12-month warranty and maintenance thatis renewable for subsequent years for an annual fee.

Software Integration

Sametime Unified Telephony 8.5.2 software integrates with:

✔ Sametime Server 8.0.2 (NNUM-7S8FXR) or later

✔ Sametime Connect Client 8.0.2 or later

✔ Lightweight Directory Access Protocol (LDAP) directory

Connections to all telephony equipment is done through a SIP trunking interface. Interfacing into theSUT system can be done through the REST APIs.




Sametime Unified Telephony embedded softphone

The Sametime Unified Telephony product includes a Sametime Connect client plug-in with anembedded softphone. Plugins and softphones are installed on the user's workstation.

SUT embedded softphones register with SIP proxy servers in the Telephony Application Servers.

• SUT embedded softphones cannot registers with any other SIP proxy server.

• Any other SIP phone, soft or hardware-based phone cannot register with SIP proxy servers in theTelephony Application Servers.

Embedded softphones communicate with other SIP phones and gateways using one of the followingcodecs:

Codec Bit rate

GIPS iSAC Adaptive and variable 10-32 kbps

iLBC 13.3 kbps (30 ms frames), 15.2 kbps (20 ms frames)

ITU-T G.729 8 kbps

ITU-T G.711 64 kbpsThe codec priority can be changed by the system administrator.

• Embedded softphones are able to mark the voice (RTP - Real-time Transport Protocol) packets withDSCP value EF to support Quality of Service (QoS). Currently the support is limited to the SametimeConnect client running on Microsoft Windows XP only.

• Embedded softphones are able to encrypt the media stream (RTP) using Secure Real-time TransportProtocol (SRTP).

• Each softphone user requires 10-100 Kbps full duplex bandwidth depending on codec used, with sub100 millisecond latency round trip and sub 10 millisecond jitter.

• SUT softphone to softphone calls use GIPS iSAC as a default codec and require up to 32 kbps bandwidth.

• Softphone to PBX and PSTN calls require 100 Kbps full duplex bandwidth (using G.711) with sub100 millisecond latency round trip and sub 10 millisecond jitter.




Telephony Application Server (TAS)

The Telephony Application Server provides telephony services to the Sametime Unified Telephonyusers through the Sametime Connect client. The Telephony Application Server interfaces withSametime Community server, enterprise directory and Telephony Control Servers.

Telephony Application Server performs the following functions:

• Runs all application logic that manages the workflow associated with routing incoming calls so thatthey are always routed to the preferred device.

• Initiates all call-related events to the client. If the Sametime Connect client is online, the TelephonyApplication Server initiates a pop-up window that lets the user answer an incoming call with thecurrent preferred device or deflect the call to another device.

• Provides personal call history records of all calls that user makes and receives.

• Handles all call and flow control for audio conferences.

• Manages all user and configuration data. Some data is provisioned into the system as part of theinitial configuration, while other data is managed by the user.

• Together with Media Server component provides limited audio conferencing capability. Primary usecase is for small ad hoc meetings. This capacity is configurable.

• Provides ability to provision remote administrative features.

• Provides SIP Registrar and Proxy service for the embedded soft phones.

There can be one or up to eight Telephony Application Servers on each SUT installation.

Each Telephony Application Server monitors up to 15,000 users. User affinity is established when a person logs onto the Sametime server with a Sametime Unified Telephony client. Once the Sametimeserver notifies the Telephony Application Server of the event, the Telephony Application Server opens achannel to the Sametime Connect client that enables the Sametime Unified Telephony features.

Users can provision as many preferred devices and phone numbers for themselves as they like. TheTelephony Application Server can ring any of these devices based on rich presence and user rules thatallow the user to determine what device they want to ring according to any combination of time,location, and presence status.

Telephony Application Servers are deployed with at least one warm standby server to provideredundancy for server failure. One warm standby server can be used to provide redundancy for one or alleight Telephony Application Servers. If higher redundancy is required, the number of warm standbyservers can be increased to match the number of Telephony Application Servers.

Telephony Application Server Specifications

• Hardware: System requirements can be found at:http://www.ibm.com/support/docview.wss?uid=swg27020284

• Operating System: SUSE Linux Enterprise Server version 10 SP3

♦ SUSE Linux Enterprise Server license can be ordered through IBM or purchased separatelyfrom Novell.

• Each Telephony Application Server can handle up to 15,000 SUT users.

IBM Sametime Unified Telephony 8.5.2 Functional S pecification 8

http://www.ibm.com/support/docview.wss?uid=swg27020284




♦ Users can provision as many telephone numbers as they like.

♦ Users can have their unified numbers associated with any SUT interconnected PBXes and/or SIP/PSTN gateways in different countries.

• Telephony Application Servers use two Gigabit Ethernet ports that IBM recommends to beconnected to two separate physical Gigabit Ethernet switches to assure high availability.

• In a production environment, the Telephony Application Servers are always deployed with at least

one warm standby server.♦ Warm standby servers are identical in configuration to the Telephony Application Servers.

➢ One warm standby server can be used to provide redundancy for 1 to 8 TelephonyApplication Servers.

➢ If higher redundancy is required, the number of warm standby servers can be increased tomach the number of Telephony Application Servers (8)

♦ Telephony Application Server failover to warm standby server will take between 20 and 40minutes on a system with 15,000 users. During this time, the system operates in a fail-safemode, meaning that the end-user routing rules are not active, but all incoming calls are routed toa device provisioned for each use as their fail safe device.

♦ Note: Small proof of concept Sametime Unified Telephony installations can be deployed withone Telephony Application Servers, but the client should be advised that one server configuration does not offer redundancy.

• Each Telephony Application Server (and warm standby server) must have two 4 Gbps Fiber HBA(Host Bus Adapter) ports for SAN (Storage Area Network) connectivity.

• Each Telephony Application Server requires a minimum of 20 GB disk space in two separate physicaldisks in dual controller SAN storage for failover support.

• Telephony Application Servers and warm standby servers require layer three IP connections to theTelephony Control Servers with following requirements:

♦ Bandwidth requirement: minimum 100 Mbps

♦ Latency: less than 50 ms roundtrip

♦ Telephony Application Servers can contain an onboard Media Server or the Media Server can be installed on a separate physical server.




Media Server

The Media Server provides announcement and ad hoc conferencing services. The announcements areavailable in different languages. The Media Server for small deployments can run inside the TelephonyApplication Server (on-board Media Server). For larger deployments requiring more voice channelcapacity, separate physical (off-board) Media Servers are recommended.


• Operating System: SUSE Linux Enterprise Server version 10 SP3.

♦ SUSE Linux Enterprise Server license can be ordered through IBM or purchased separatelyfrom Novell.

• Off-board Media Servers are paired with Telephony Application Servers.

♦ There can be a maximum of eight Media Servers per Sametime Unified Telephony installation paired with Telephony Application Servers.

♦ In addition, there can be one off-board Media Server that is dedicated for announcements.

• Each off-board Media Server must have two 4 Gbps Fiber ports for SAN storage connectivity.

• Each off-board Media Server requires 10GB disk space on each of two separate physical disks in dualcontroller SAN storage for failover support.

• The voice over IP media (RTP - Real-time Transport Protocol) traffic between SUT Media Server and interconnected IP PBXes, SIP gateways and SIP endpoints requires 100 Kbps (with protocoloverhead) full duplex bandwidth per voice channel (using G. 711) with sub 100 millisecond latencyround trip and sub 10 millisecond jitter.

• SUT Media Servers with today's design guidelines are deployed either as integrated inside theTelephony Application Server or as a separate physical server paired with TAS servers. Each MediaServer provides one logical conference bridge.

• If the Media Server is deployed as a separate physical server, the conference bridge inside the MediaServer node can handle up to 500 x G.711 and 160 x G.729 media channels. A media channel is anactive call with an active RTP based media stream.

• Each SUT ad hoc conference participant requires 100 Kbps full duplex bandwidth per voice channel(using G.711) with sub 100 millisecond latency round trip and sub 10 millisecond jitter.

• Off-board Media Servers require layer three IP connections with Telephony Application Servers andTelephony Control Servers with following requirements:

♦ bandwidth requirement: minimum 100 Mbps

♦ latency: less than 50 ms roundtrip

Media Server Conferencing capability• Media Servers support ad hoc conferencing. Users can initiate conference calls and the conference

bridge in the Media Server dials out to participants.

♦ Once the ad hoc conference has been started by the user, the conference bridge in the MediaServer can also accept inward dialing into the conference bridge. The conference leader can seethe dial-in number and passcode.






• An onboard Media Server can handle up to 75 concurrent conference call legs. For example, there can be 18 conferences with 4 people each, or 15 conferences with 5 people, and so on. Using standarderlang calculations of conference usage, and taking into account that these are ad hoc conferences (notscheduled "meet me" conferences), this amount is expected to cover 7,500 users per TelephonyApplication Server assuming a 1% adhoc conference workload. Above this 7,500 number, it isrecommended to move to an off-board Media Server deployment. It is also recommended to startwith an off-board deployment if you expect to have more than 7,5000 users in the future.

Note One erlang is the equivalent of one call, including call attempts and holding time, in a specificchannel.

• Media Server supports following codecs: G.711 A/U and G.729 AB

• An off-board Media Server can handle up to

♦ 500 simultaneous conference call legs with G.711 codec.

♦ 160 simultaneous conference call legs with G. 729 codec.

• Media Server can transcode between: G.711 A/U and G.729 AB

• There is no fixed limit to the number of simultaneous ad hoc conferences other than that the limit is

based on the maximum number of total conference call legs (participants) per Media Server.• Default number of ad hoc participants per conference is six ( 6 )

♦ The default number of ad hoc participants per conference can be modified by the administrator

♦ The default number of ad hoc participants per conference applies for all Media Servers on an SUTcluster

• The Sametime Unified Telephony users are associated with one particular Telephony ApplicationServer and its accompanying Media Server, and use the conference bridge in that particular MediaServer to initiate ad hoc conferences.

• Each Media Server contains one logical ad hoc conference bridge.

• The conference bridge dials whatever number the user enters into the "Invite Others..." window.

• If the user drag-and-drops a contact, the conference bridge uses the contact's configured phonenumber in the directory or the number the user has entered for the contact in their phonebook if it is a personal contact - in that exact format

• Each onboard and off-board Media Server contains one logical ad hoc conference bridge that has oneassigned phone number, which it uses to dial out to conference participants and to receive incomingDirect Inward Dial(DID) calls.

• Logical ad hoc Conference Bridge can dial participants in different locations and countries.

♦ When dialing a number external to PSTN or mobile numbers, the SUT conference bridge presentsits own assigned phone number as the originator of the call. If the call is routed through the IP

network to break out to PSTN or mobile network in another country, the number must bemodified at the PBX or PSTN gateway to a valid E.164 number for that country.

♦ Same technique, modifying the caller number at the PSTN breakout point is used with SUT least-cost routing design.

♦ Calling in an ad hoc conference call is only possible using the number assigned to the conference bridge, as this is what is visible to the conference leader. In the international setting, participantswanting to call in from other countries must make an international call.




Telephony Control Server (TCS)

Telephony Control Server is a Back-to-Back User Agent (B2BUA) that handles SIP signaling betweenPBXes and PSTN.

IP PBXes are connected to Telephony Control Server through SIP trunks. Non-IP PBXes and PSTNconnections are connected through SIP gateways.

Telephony Control Server is always deployed in a duplex configuration with RAID hard disk drives to provide high availability.

A pair of Telephony Control Servers supports up to 100,000 subscribers. The Telephony Control Server hosts all subscriber data (unified numbers) for the entire user population.

The Telephony Control Server is responsible for receiving incoming requests. The Telephony ControlServer also provides the unified number service and handles all incoming call routing for the unifiednumber. At a conceptual level, the Telephony Control Server container is a back-to-back user agent(B2BUA) with a PBX abstraction layer that allows it to work with Telephony Application Server.

B2BUA's inherent “two-call-legs approach” allows Sametime Unified Telephony to provide PBX-likefunctions that operate homogeneously across any vendor's PBX and facilitate an integratedheterogeneous UC2 solution. To provide this B2BUA functionality across the enterprise, the TelephonyControl Server maintains a SIP signaling path between the telephony service equipment. The SIP trunksare established to other equipment via configuration settings on the Telephony Control Server that allowSametime Unified Telephony to process incoming calls from various trunks and/or route calls to varioustrunks according to the numbering plan configuration. Each trunk is configured as a SIP end point.

Telephony Control Server Specifications


The part number specified in the system requirements is the only part number that can be used as a

Telephony Control Server. The Sametime Unified Telephony Connect software can not be installed toany other server.

The specified part number in the system requirements document comes with a predefined set of adapters and configuration, including two 300 GB SAS hard drives configured into RAID and withtwo redundant power supplies.

Use of this specified part number ensures that all the correct components are shipped with the server and that the configuration will match what has been tested by IBM software development and performance test teams.

• Software: Sametime Unified Telephony Connect component of Sametime Unified Telephony.

• Telephony Control Server is always deployed with two servers that enable server failover and veryhigh availability.

• Telephony Control Server failover is performed in 10-45 seconds depending on the conditions of thefailure.

• Telephony Control Servers have eight Gigabit Ethernet ports and require two sets of physical Ethernetswitches with three Virtual LANs (Billing, Management and Signaling) and two direct server-to-server interconnect cables to assure high availability.






• Two Telephony Control Servers must be on the same layer two subnet with the followingrequirements:

♦ The network links used for interconnects must have low latency and low error rates.

♦ The interconnects should not be used on any network that might experience network outages of 5 seconds or more.

• Telephony Control Server can handle up to 100,000 users.

• Telephony Control Server can handle up to 15,000 SIP trunks.

• Telephony Control Server acts as SIP B2BUA between PBXes (IP PBXes and Non IP PBXes throughSIP gateways).

♦ Telephony Control Server can connect to the PSTN through SIP gateways

♦ Telephony Control Server is connected to IP PBXes, non-IP PBXes and gateways through SIPtrunks.

• Telephony Control Server can support up to eight active Telephony Application Servers and eightwarm standby servers.

♦ The number of users can not exceed 100,000 users.

SUT can be deployed in various geographical models, depending on the redundancy requirements of thedeployment. It is possible to install SUT between the data centers, provided the right level of network support exists between the centers.




Storage Area Network

Warm standby servers, external storage, Fiber Channel Storage Area Network and IBM Tivoli SystemAutomation for Multi-platform (SAMP) are used to provide failover support for Telephony Applicationand Media Servers.

Telephony Application Servers, Media Servers and warm standby servers are connected to external

storage device through 4 Gbps Fiber Channel Storage Area Network switches.To increase redundancy the Telephony Application Servers and Media Servers should have HBAadapters with two 4 Gbps Fiber Channel ports connected to two parallel Fiber Channel SAN switches.

IBM Tivoli System Automation for Multi-platform (SAMP) software in the Telephony Application andMedia Servers provides server failover automation and reduces the duration of Sametime UnifiedTelephony system outage. SAMP automatically switches users, resources and applications from afailing server to warm standby server after software or hardware failure.

Storage Area Network Specifications

• A 4 Gbps Fiber Channel switch can be used if it is compatible with QLE2462 based 4G Fiber

Channel Dual Port PCIe HBA (Low profile) card.

• Tested hardware: 249824E – IBM System Storage SAN24B-4 Express.

• For improved redundancy, IBM recommends deployment of two parallel 4 Gbps Fiber Channelswitches.

Network Storage Specifications

External network storage system contains data and binaries for Telephony Application and MediaServers. Two virtual disks are required for each Telephony Application and Media Server.

• A SAN solution using a compatible QLE2462-based 4G Fiber Channel Dual Port PCIe HBA (Low profile) card can be used with Sametime Unified Telephony.

• Tested hardware: IBM System Storage DS3400.

♦ DS3400 – 1726-42X -dual controller with 4 Gbps Fiber Channel (FC) interface

• Each Telephony Application and off-board Media Server requires 20 GB disk space and 10 GBrespectively on two separate physical disks. For a maximum deployment scenario, with sevenTelephony Application Servers and seven Media Servers, 210 GB of disk space is required.

♦ IBM System Storage DS3400 supports combination of 12 SAS or SATA 3.5" drives per enclosure.

♦ IBM System Storage DS3400 is scalable to 5.4 TB of storage capacity with 450 GB hot-swappable SAS disks or 12.0 TB with 1.0 TB hot-swappable SATA disks in the first enclosure.

♦ IBM System Storage DS3400 is expandable by attaching up to three EXP3000s for a total of 21.6 TB of storage capacity with 450 GB SAS or up to 48.0 TB with 1.0 TB SATA.

See the Sametime Unified Telephony product pages for list of interoperability-tested network storagesolutions.




IP Network Connections

All SUT servers are equipped with Gigabit Ethernet ports for server-to-server and external systemsinterconnection. If the SUT servers are co-located in same data center, a Gigabit (or at a minimum 100Mbps) Ethernet switches are required to assure sufficient interconnection bandwidth and latency.

IBM recommends connecting the SUT servers to two sets of parallel, cross-connected Ethernet switchesto improve network redundancy. This method of interconnection prevents system outage in case of

Ethernet port, Ethernet switch or Ethernet interconnect cable failure.

The TAS interconnection to the Sametime Community server and LDAP directory is recommended to bethrough two redundant parallel IP network connections to assure high availability.

• Telephony Application Servers and Sametime Community server exchange phone presence andcall/conference call setup, transfer and tear down signaling using bandwidth efficient VP (VirtualPlaces) protocol.

• Telephony Application Servers, the Sametime Community server and LDAP directory server do nothave to be co-located as long as the bandwidth and latency requirements are met.

• Telephony Application Servers and LDAP directory connection is used for occasional user directoryupdates and is not considered bandwidth critical.

The amount of SIP signaling traffic between SUT embedded softphones and Telephony ApplicationServers, and signaling between Telephony Application Servers, Media Servers, Telephony ControlServer and PBXes and SIP gateways depends on the number of users on the system and how many callsthey make.

• Although this is signaling traffic, rather than voice media traffic, the IP network needs to providesufficient network bandwidth and latency to handle the traffic.

• The signaling and data traffic between TCS, TAS and Media Servers, as well as the data traffic between TAS server and Sametime Community server and enterprise directory requires 100 Mbps bandwidth and sub 100 millisecond latency.

SUT users can initiate phone calls and conference calls using:

• SUT embedded softphones

• IP phones connected to IP PBXes

• Non-IP phones connected to non-IP PBXes

• PSTN phones

• Mobile phones

Voice traffic between SUT embedded softphones and IP phones, as well as the traffic from them to theMedia Server and voice gateways travels over the enterprise IP network and the network needs to be ableto support RTP media streams and Quality of Service.

Although the SUT embedded softphones are able to mark the voice packets for priority, the remote users

accessing organization’s telephony system over VPN and Internet connections may experienceoccasional deterioration of voice quality as the Internet is not able to assure the quality of service.

All SUT conference call traffic travels over the enterprise IP network as the SUT Media Server acts as aconference bridge joining the calls from all phones (IP and non-IP) on the conference.

• Non-IP, PSTN and mobile phones join conference calls through SIP gateways.

• Voice traffic for non-IP, PSTN and mobile phones runs over the enterprise IP network if the other party is an SUT softphone or a phone on non-IP PBX connected through SIP gateway.




IP Network Specifications

• Telephony Application Servers, Media Servers and warm standby servers have two Gigabit Ethernet ports.

♦ IBM recommends connecting the ports to two physical, but cross-connected Ethernet switchesto improve network redundancy.

• Telephony Control Servers have eight Gigabit Ethernet ports.

♦ Two of the ports are used for direct server-to-server interconnect cables.

♦ The other six ports need to be connected to either two physical, but cross-connected Ethernetswitches with virtual LAN (VLAN) capability or to two sets of three cross connected physicalEthernet switches.

♦ billing, management and signaling ports need to be connected to separate Ethernet switches or VLANs.

♦ Telephony Control Servers exchange signaling with:

➢ Interconnected PBXes and SIP gateways through SIP trunks - (call setup, transfer and tear down)

➢ Telephony Application Servers for call setup, transfer and tear down.

➢ Telephony Control Servers for media gateway functionality.

• Telephony Application Servers exchange signaling with:

♦ Sametime Community server for phone presence and call/conference call setup, transfer andtear down.

♦ Media Servers for conference call setup, transfer and tear down.

♦ Embedded softphones at end user workstations for call setup, transfer and tear down.

• Media Servers exchange signaling with:

♦ Telephony Application Servers for conference call setup, transfer and tear down (see above)

• Media Servers have RTP media connections with:

♦ SUT embedded softphones for announcements and conference calls.• required bandwidth: 80 kbps per softphone (to cover G.711 codec).

• maximum latency: 100 ms roundtrip

♦ IP phones and SIP gateways for announcements and conference calls.

• required bandwidth: 80 kbps per IP phone or gateway (to cover G.711 codec).

• maximum latency: 100 ms roundtrip average – 200 maximum roundtrip

• Embedded softphones in end user workstations have RTP media connections with:

♦ Other SUT embedded softphones

♦ IP phones

♦ SIP gateways♦ Media Servers

➢ Required bandwidth varies depending on the codec negotiated between the endpoints. Tocover G.711 codec, allocate 80 kbps bandwidth for each voice call.

➢ Maximum latency: 100 ms round-trip




Interoperability Testing Information

Sametime Unified Telephony ("SUT") software is designed to connect to two classes of PBXes:

• SIP-compliant PBXes and Gateways, using relevant SIP standards (Primary RFCs: 3261, 3264, and4566)

• TDM (Time Division Multiplex) telephone systems via SIP gateways that have been tested and

release by third party suppliers

SUT connects to the PSTN (Public Switched Telephony Network) via the connected PBXes or SIPGateways that have been tested and released by third party suppliers. Direct SIP trunking may beavailable from certain carriers.

Customer projects can have unique characteristics, and need to be verified as part of their deployment,which may result in configuration changes to existing PBXes, Gateways, and/or SUT.

In response to widespread industry support for SUT, IBM has launched a testing program for the product. This unified communications program allows IBM and key Business Partners to test their product's capabilities with SUT. This testing helps ensure that customers will be able to successfullyimplement SUT into their existing IT environment. The growing list of participating suppliers can befound in the Sametime Unified Telephony 8.5.2 System Requirements at:http://www.ibm.com/support/docview.wss?uid=swg27020284

For vendors and partners who want to engage in this testing program, information can be found at:http://www-01.ibm.com/software/lotus/products/sametime/unifiedtelephony/features/



http://www-01.ibm.com/software/lotus/products/sametime/unifiedtelephony/features/






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