IBM Lotus Sametime Unified Telephony functional specification

IBM Lotus Sametime Unified Telephony functional specification

IBM Lotus Sametime Unified Telephony Functional Specification

Contents

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

Features................................................................................................................................................3

Benefits .................................................................................................................................................3

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

Summary of Administrative Features........................................................................................................4

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

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

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

Telephony Application Server Specifications........................................................................................7

The Media Server Conferencing capability ...........................................................................................9

Telephony Control Server Specifications............................................................................................11

Storage Area Network Specifications .................................................................................................12

Network Storage .....................................................................................................................................12

Network Storage Specifications ..........................................................................................................12

IP Network Connections .........................................................................................................................13

IP Network Specifications ...................................................................................................................14

Interoperability Testing Information.........................................................................................................15

© Copyright IBM Corporation 2009, 2010. All rights reserved. 2

IBM Lotus Sametime Unified Telephony Functional Specification

Lotus Sametime Unified Telephony:

Functional Specification

IBM

®

Lotus

®

Sametime

®

Unified Telephony (SUT) software is middleware that integrates a market- leading IBM

Lotus Sametime unified collaboration solution with a telephony solution across multi-vendor Private Branch

Exchange (PBX) systems and provides a unified end user experience from either inside the Lotus 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 leveraging and extending the existing telephony solutions and unifying them with real-time collaboration tools, an organization can help users find and reach each other more easily and collaborate working together smarter to speed decisionmaking 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 functionality from 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 call management 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) based telephony. The software integrates both IP-based and legacy-based PBXs.

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

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IBM Lotus Sametime Unified Telephony Functional Specification

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, along with online presence status (Available, Away, In a meeting, Do not disturb, and so forth), making it easy to know whether it is appropriate to initiate a phone conversation. Users can see if the person they want to contact is on the phone or off the phone.

Unified business number – Users have a single unified phone number, but the Sametime Unified

Telephony 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 Lotus Sametime software provides inherent presence and location awareness, SUT software can automatically set the preferred device based on the user's availability and location status.

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

Call control - Users are able to manage their phone calls through the Sametime Unified Telephony's visual call control interface regardless whether they use softphone, desk or mobile phone. Users can see the call and connection status, callers business card, and have call controls that enable pausing and resuming the call or transferring the call to another phone device. Users can also forward the call to another person or to terminate the call. The call control interface also allows initiating other means of 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 names from the Sametime contacts list or QuickFind, search the contact from the enterprise directory. Users may 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 who is 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 so forth.

Softphone – Users can use their workstation's built-in audio capability or attached headsets to initiate and 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 to enable one-click conference calling, including conference call passwords.

Personal call history – All Sametime connected calls are logged into a user's personal call history log.

Calls from the log can be added to the user's phonebook, re-dialed or added to the user's contact list.

Summary of Administrative Features

Add or remove users or subscribers to Sametime Unified Telephony.

Set up secure communication for IBM Lotus Sametime Unified Telephony communities.

Monitor call activity and general server status.

Set up SIP proxy and registrar properties.

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IBM Lotus Sametime Unified Telephony Functional Specification

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. Each area has a different set of languages that is supports:

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

Conferences contain the audio messages used during the conference life cycle for the conference management: 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 call management: Argentines, Bulgarian, Czech, Danish, German, German Austria, Greek, English Australia,

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 with telephony 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 telephony system.

The Sametime Unified Telephony solution consists of:

• IBM Lotus 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 PBXs and Gateways (SIP trunks) o

to Sametime Unified Telephony softphones

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

© Copyright IBM Corporation 2009, 2010. All rights reserved. 5

IBM Lotus Sametime Unified Telephony Functional Specification

Sametime Unified Telephony Components

IBM Lotus Sametime Unified Telephony 8 software

PN 5724U79 contains two components:

IBM Lotus Sametime Unified Telephony Call - provides Telephony Application Server (TAS) and

Media Server (MS) software. Sametime Unified Telephony Call includes embedded softphones.

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

Software Integration

IBM Lotus Sametime Unified Telephony 8 software integrates with:

IBM Lotus Sametime Server 8.0.2 (NNUM-7S8FXR) or later

IBM Lotus 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 the SUT 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 an embedded 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 the

Telephony Application Servers.

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

codec bitrate

GIPS iSAC iLBC

ITU-T G.729

ITU-T G.711

Adaptive and variable 10-32 kbps

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

8 kbps

64 kbps

The codec priority can be changed by the system administrator.

Embedded softphones are able to mark the voice (RTP - Real-time Transport Protocol) packets with DSCP value EF and signaling with CS3 to support Quality of Service (QoS).

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IBM Lotus Sametime Unified Telephony Functional Specification

Embedded softphones are able to encrypt the media stream (RTP) using Secure Real-time Transport

Protocol (SRTP).

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

Telephony Application Server (TAS)

The Telephony Application Server provides telephony services to the Sametime Unified Telephony users through the Sametime Connect client. The Telephony Application Server interfaces with Sametime 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 that they are always routed to the preferred device.

Initiates all call-related events to the client. If the Sametime Connect client is online, the Telephony

Application Server initiates a pop-up window that lets the user answer an incoming call with the current 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 the initial configuration, while other data is managed by the user.

Together with Media Server component provides limited audio conferencing capability. Primary use case 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 Sametime server notifies the

Telephony Application Server of the event, the Telephony Application Server opens a channel 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. The Telephony

Application Server can ring any of these devices based on rich presence and user rules that allow 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 provide redundancy for server failure. One warm standby server can be used to provide redundancy for one or all eight Telephony

Application Servers. If higher redundancy is required, the number of warm standby servers can be increased to match the number of Telephony Application Servers.

Telephony Application Server Specifications

Hardware: IBM System x3550 M2 with following configuration:

794692U – Rack mounted x3550 M2, Xeon 4C X5570 95W 2.93GHz/1333MHz/8MB L3, 2x2GB, O/Bay

2.5in HS SAS/SATA, SR BR 10-I, CD-RW / DVD-ROM Combo UltraSlim Enhanced with 675W power supply,

46M1087 – Second Intel Xeon 4C Processor Model X5570 95W 2.93GHz/1333MHz/8MB

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IBM Lotus Sametime Unified Telephony Functional Specification

44T1481 – two 2GB Dual Rank PC3-10600 CL9 ECC DDR3 LP RDIMM 1333MHz

42D0632 – two IBM 146 GB 2.5in SFF Slim-HS 10K 6Gbps SAS HDDs configured to RAID

39R6527 – QLogic 4Gb FC Dual-Port PCIe HBA for IBM System x

46M1075 – Second redundant 675W Power supply

Operating System: SUSE Linux Enterprise Server version 10 SP2

SUSE Linux Enterprise Server license can be ordered through IBM or purchased separately from

Novell.

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

Users can provision as many telephone numbers as they like.

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

Telephony Application Servers use two Gigabit Ethernet ports that IBM recommends to be connected 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 Telephony Application

Servers.

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

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

Note: Small proof of concept Sametime Unified Telephony installations can be deployed with one

Telephony Application Servers, but the client should be advised that one server configuration does not offer redundancy.

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

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.

Telephony Application Servers and warm standby servers require layer three IP connections to the

Telephony 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.

© Copyright IBM Corporation 2009, 2010. All rights reserved. 8

IBM Lotus Sametime Unified Telephony Functional Specification

The Media Server

The Media Server provides announcement and ad-hoc conferencing services. The announcements are available in different languages.

The Media Server for small deployments can run inside the Telephony Application Server (on-board Media

Server). For larger deployments requiring more voice channel capacity, separate physical (off-board) Media

Servers are recommended.

Hardware: An off-board (separately-installed) Media Server has the same physical configuration as the

Telephony Application Server: IBM System x3550 M2.

Operating System: SUSE Linux Enterprise Server version 10 SP2

SUSE Linux Enterprise Server license can be ordered through IBM or purchased separately from

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 dual controller SAN storage for failover support.

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

SUT Media Servers with today's design guidelines are deployed either as integrated inside the Telephony

Application Server or as a separate physical server paired with TAS servers. Each Media Server provides one logical conference bridge.

If the Media Server is deployed as a separate physical server, the conference bridge inside the Media

Server node can handle up to 500 x G.711 and 160 x G.729 media channels. A media channel is an active 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 and

Telephony Control Servers with following requirements:

♦ bandwidth requirement: minimum 100 Mbps latency: less than 50 ms roundtrip

The 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 Media Server can also accept inward dialing into the conference bridge. The conference leader can see the 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 standard erlang calculations of conference usage, and taking into account that these are ad-hoc conferences (not

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IBM Lotus Sametime Unified Telephony Functional Specification scheduled "meet me" conferences), this amount is expected to cover 7,500 users per Telephony

Application Server assuming a 1% adhoc conference workload. Above this 7,500 number, it is recommended to move to an off board media server deployment. It is also recommended to start with an off-board deployment if it is expected that the number of users on a TAS will be higher than 7,500 at some point in the future.

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

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 SUT cluster

The Sametime Unified Telephony users are associated with one particular Telephony Application Server and its accompanying Media Server, and use the conference bridge in that particular Media Server 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 phone number 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 one assigned phone number, which it uses to dial out to conference participants and to receive incoming Direct

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 presents its 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 the mobile network in another country, the number must be modified 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 into 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 this means that participants wanting to call in from other countries would need to make an international call.

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IBM Lotus Sametime Unified Telephony Functional Specification

Telephony Control Server (TCS)

Telephony Control Server is a Back-to-Back User Agent (B2BUA) that handles SIP signaling between PBXs and

PSTN.

IP PBXs are connected to Telephony Control Server through SIP trunks. Non-IP PBXs and PSTN connections are connected through SIP gateways.

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

99.999% 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 Control Server also provides the unified number service and handles all incoming call routing for the unified number. 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-like functions that operate homogeneously across any vendor's PBX and facilitate an integrated heterogeneous UC2 solution. To provide this B2BUA functionality across the enterprise, the Telephony Control Server maintains a SIP signaling path between the telephony service equipment. The SIP trunks are established to other equipment via configuration settings on the Telephony Control Server that allow Sametime Unified Telephony to process incoming calls from various trunks and/or route calls to various trunks according to the numbering plan configuration. Each trunk is configured as a SIP end point.

Telephony Control Server Specifications

Hardware: IBM System x3650T with specific part number: 79805CX

79805CX is the only part number that can be used as a TCS server. The IBM Lotus Sametime Unified

Telephony Connect software can not be installed to any other server.

This part number comes with a pre-defined set of adapters and a configuration, including two 146 GB

SCSI hard drives configured into RAID and with two redundant power supplies.

Use of this specified part number insures 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.

Operating System: SUSE Linux Enterprise Server version 9. OS is included in the IBM Sametime Unified

Telephony software bundle.

Software: IBM Lotus Sametime Unified Telephony Connect component of Sametime Unified Telephony.

Telephony Control Server is always deployed with two physical x3650T (P/N 79805CX) servers that enable server failover and 99.999% availability.

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

Telephony Control Servers have eight Gigabit Ethernet ports and require two sets of physical Ethernet switches with three Virtual LANs (Billing, Management and Signaling) and two direct server-to-server interconnect cables to assure 99.999% availability.

Two Telephony Control Servers must be on the same layer two subnet with the following requirements:

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.

© Copyright IBM Corporation 2009, 2010. All rights reserved. 11

IBM Lotus Sametime Unified Telephony Functional Specification

Telephony Control Server acts as SIP B2BUA between PBXs (IP PBXs and Non IP PBXs through SIP gateways).

Telephony Control Server can connect to the PSTN through SIP gateways

Telephony Control Server is connected to IP PBXs, non-IP PBXs and gateways through SIP trunks.

Telephony Control Server can support up to eight active Telephony Application Servers and eight warm 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 the deployment. 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 System Automation for Multi-platform (SAMP) are used to provide failover support for Telephony Application and 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 HBA adapters 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 and Media

Servers provides server failover automation and reduces the duration of Sametime Unified Telephony system outage. SAMP automatically switches users, resources and applications from a failing server to warm standby server after software or hardware failure.

Storage Area Network Specifications

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

For improved redundancy, IBM recommends deployment of two parallel 4 Gbps Fiber Channel switches.

Other vendor’s 4 Gbps Fiber Channel switches can be used if they are compatibility tested with: 39R6527 –

QLogic 4 Gbps FC Dual-Port PCIe HBA for IBM System x.

Network Storage

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

Network Storage Specifications

Tested hardware: IBM System Storage DS3400.

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

Each Telephony Application and off-board Media Server requires 20 GB disk space and 10 GB respectively on two separate physical disks. For a maximum deployment scenario, with seven Telephony 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.

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IBM Lotus Sametime Unified Telephony Functional Specification

Other vendor’s 4 Gbps Fiber Channel network storage solutions can be used once interoperability tested with Sametime Unified Telephony.

See the Sametime Unified Telephony product pages for list of interoperability tested network storage solutions.

IP Network Connections

All SUT servers are equipped with Gigabit Ethernet ports for server to server and external systems interconnection.

If the SUT servers are co-located in same data center, a Gigabit or at a minimum 100 Mbps 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 switches to improve network redundancy. This method of interconnection prevents system outage in case of Ethernet port,

Ethernet switch or Ethernet interconnect cable failure.

Telephony Application Servers ' interconnection to the Sametime Community server and LDAP directory is recommended to be through two redundant parallel IP network connections to assure high availability.

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

Telephony Application Servers, the Sametime Community server and LDAP directory server do not have 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 directory updates and is not considered bandwidth critical.

The amount of SIP signaling traffic between SUT embedded softphones and Telephony Application Servers, and signaling between Telephony Application Servers, Media Servers, Telephony Control Server and PBXs and SIP gateways depends on the number of users on the system and how many calls they make.

Although this is signaling traffic, rather than voice media traffic, the IP network needs to provide sufficient 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 PBXs

Non-IP phones connected to non-IP PBXs

PSTN phones

Mobile phones

Voice traffic between SUT embedded softphones and IP phones, as well as the traffic from them to the Media

Server and voice gateways travels over the enterprise IP network and the network needs to be able to 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 experience occasional 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 a conference 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.

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IBM Lotus Sametime Unified Telephony Functional Specification

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 switches to improve network redundancy.

Telephony Control Servers have eight Gigabit Ethernet ports.

Two of the ports are used for direct server-to-server interconnect cables. other six ports need to be connected to either two physical, but cross-connected Ethernet switches with virtual LAN (VLAN) capability or to two sets of three cross connected physical Ethernet switches.

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

Telephony Control Servers exchange signaling with:

Interconnected PBXs 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:

Lotus Sametime Community server for phone presence and call/conference call setup, transfer and tear 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. To cover

G.711 codec, allocate 80 kbps bandwidth for each voice call.

Maximum latency: 100 ms roundtrip

© Copyright IBM Corporation 2009, 2010. All rights reserved. 14

IBM Lotus Sametime Unified Telephony Functional Specification

Interoperability Testing Information

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

• SIP-compliant PBXs and Gateways, using relevant SIP standards (Primary RFCs: 3261, 3264, and 4566)

• TDM (Time Division Multiplex) telephone systems via SIP gateways that have been tested and release by third party suppliers

Lotus SUT connects to the PSTN (Public Switched Telephony Network) via the connected PBXs or SIP Gateways that have been tested and released by third party suppliers. Direct SIP trunking may be available 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 PBXs, Gateways, and/or SUT.

In response to widespread industry support for Lotus 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 successfully implement SUT into their existing

IT environment. The growing list of participating suppliers can be found at: http://www-01.ibm.com/support/docview.wss?&uid=swg27015331

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/

© Copyright IBM Corporation 2009, 2010. All rights reserved. 15

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