Whats New in SIP Server 8...Integration with Softswitch supported (Siemens, Cisco & others) Traffic...

What’s New in SIP Server 8.1

Marc Sassoon Genesys University, Master Technical Instructor September 2012

©2012, Genesys Telecommunications Laboratories – All Rights Reserved 2

What is Genesys IP Solution?

SIP Server

GVP & Media Server

UC Connect


Genesys SIP Contact Center

Virtualize Customer Services

• Virtualize CC operation • Intelligent routing • Unified mgmt & reporting

• Open & standards-based • Foster service innovations • Investment protection

Innovate for Growth

• No CC-PBX or ACD required! • 100% software • IT friendly solution

Simplify CC Infrastructure

ACD CC-PBX


Genesys SIP - Market Leader by Gartner

“In 2011, Genesys sold more

solutions based on its SIP Server

platform, capable of supporting a

"PBX free" environment for

contact centers, than those based

on its traditional T-Server

platform, indicating the company's

strengthening performance in

selling its emerging technologies.”

- Gartner, June/2012 -


UC

Connect

Genesys SIP – Transforming the Contact Center

Total Shipments

300,000+

Total Customers

500

SIP

Voicemail

Business

Continuity

99.999%

Availability

SIP

Ecosystem

Active

Recording


IP Solution overview (1 of 2)

SIP Application Server focused on Contact Centers Stand Alone Contact Center

Soft Switch Integration (SIP-based T-Server)

IMS Application Server

Flexible agent deployment models IP-based ACD Platform

Branch Office, Remote/Home Agent, Expert Agent

Outsourced Contact Center

Hosted Services

Multiple types of media – IM, Voice, Video


SIP Agents

PSTN

VoIP

Gateway

SIP

Server Media

Server

Customers IP Network

TDM Trunk

SIP Trunk

SIP Server Summary

Full feature ACD Platform

Integration with Genesys routing, reporting, agent desktop

Seamless migration from TDM environments

Integration with Softswitch supported (Siemens, Cisco & others)

Traffic Distribution by Network SIP Server

Full Support for Remote Agents and Nailed-up connections

Open, Standards-based SIP

Genesys SIP Solution – GVP, Outbound/Proactive, QM/CR, SIP Endpoint

Compatible with Framework 7.6, 8.0 & 8.1 (but some newer features require newer Fwk)

7

Genesys

CIM

Network SIP

Server

Genesys

QM/CR


IP Solutions overview (2 of 2)

Part of the Genesys Open Systems Layer Standards-based SIP Signaling “replaces” CTI Link

Full integration with Genesys Routing & Reporting Solutions

Integrated with GVP

Agent & Telephony level features

First and Third Party Call Control

Interfaces with Media Server or SM

Music on hold/queue, Announcements

Conference, Monitoring, Recording

Multiple types of Media

Voice, Video, IM


Components

SIP Server Genesys T-Server and SIP softswitch

Media Server Media Server and Multipoint Conference Unit (MCU)

Network SIP Server Programmable load-balancer (SIP Redirect Server)

9


Genesys SIP Server provides an IP-based solution offering of all Genesys capabilities:

SIP Server is a T-Server

SIP Server is IP PBX for CC

SIP Server can act as a SIP Softswitch

Support for all Genesys solutions

Support for SIP and TDM agents

100% Software

SIP Server serves role as:

Back to Back User Agent (B2BUA)

SIP Application Server

SIP Registrar

Multiple Operating System Support

Microsoft Windows, IBM AIX, Sun Solaris, Linux

Scalable & Robust

From 10 agents to 30K(tested capacity) agents per system. Hot Standby (HA) & Disaster Recovery

Genesys SIP Server


Basic architecture - standalone


Basic architecture – application server mode


Media Server

Unification of media processing capabilities GVP Media Control Platform

Stream Manager

Third Party (HMP)

Will provide media services to Genesys Solutions GVP (VoiceXML, CCXML)

IP Solution (SIP Server, Cisco T-Server)

Outbound IP (Proactive Notification, Outbound Contact)

Performance bar Far better than SM

Works with GVP Resource Manager Load balancing, scaling, policy enforcement

13


Sizing and Scaling SIP Server Scaling

SIP Server SIP Server supports full hot

standby Up to 150 direct calls per sec per

instance of SIP Server Up to 15,000 End points monitored

per instance of SIP Server

Load balancing with Network SIP Server Use of Network SIP Server to load

balance over multiple instances of SIP Server

Network SIP Server can handle up to 400 calls per sec (based on Redirect)

Global scalability Tested to 30,000 agents Potentially can scale much higher


Network SIP Server

Used to load balance between multiple SIP Servers

Tested up to 4000 calls/sec for each Network SIP Server

Solution tested (SIP Server and Network SIP Server together) to 30,000 agents. Conceivably could go much higher.

Gateways can load balance between Network SIP Servers

Very reliable and simple

18

General Features & Benefits


Functional enhancement

ACD subsystem Mapping phone state to an agent state in Genesys

Stranded call handling

NAS for agent

Nailed up agent New extension of remote agent configuration

Supervision Recording of the consultation part of the call

Supervision for transferred calls

20


Functional enhancement

Early media Support for early media for call arriving to routing points

Music on hold Flexible configuration on per DN or even per call/event basis

Greeting Advanced configuration options allowing to change the way SIP Server

applies logic for customer/agent greetings

Benefits: Customer experience

Agent productivity

21


New enhancements

Media Server protection Automatic restart of recording, conferences, treatments

Enhanced gateway selection Transparent selection of subsequent MGW in case of failure

Overload protection Constant monitoring of 3 vital SIP stack parameters

QoS support Different packet mark for each traffic type to allow network

switches/routers to apply priorities and policies

Benefits: Stable and reliable platform providing continuous service for agent and

customers

22

Genesys SIP 5x9 Availability

99.999%

Template Version 1.1


5x9 (99.999%) availability is achieved through the combination of –

• Superb quality solution

• High availability architecture and configuration

• Best practices around organization, operation, and management

Genesys SIP solution availability measurement is through –

• Robustness Testing

• Failure Rate Prediction

• Platform Availability Analysis

• Customer Data

Genesys 5x9 Availability – NOT Function/Feature


SIP Server 8.1 - Detailed capacity information

Call Flow SIP Server Performance

Multi-threaded Configuration

Simple ~ 150 cps

Moderate 70 - 100 cps

Complex 20 – 60 cps

SIP Server Capacity

Windows 2008 64-bit

Red Hat 5 64-bit

Others

Registered Agents (with “aggregated” T-Lib Connection)

15,000 15,000 10,000

Registered Agents (each with “direct” T-Lib Connection)

15,000 4,000 4,000

Simultaneous Calls 15,000 15,000 4,500

Why would a customer need Genesys IP Solution?


SIP Server All IP Environment

PSTN

SIP CC Agents Home Agent Branch Office Agents

SIP Value Prop:

Toll By Pass

All Open SIP Standard

SIP applications available

for all agents

SoHo

SIP Server

SIP

Customers

Stream Manager/

Media Server


Branch Office Deployment with SIP Server

SIP S

IP (QoS) network

SIP Agents – Remote, Knowledge workers, outsource

SIP Value Prop:

Extension to branch w/ No

ACD or PBX at branch

All software based

Open SIP Standard

Light footprint

T-Server

Stream Manager

/

Media Server

Branch office

Soho

SIP CC Agents Enterprise TDM Agents

PSTN

VoIP Gateway

Legacy PBX


Home Agent – Support for IP or TDM

SIP Value Prop:

No CTI

Traditional TDM home

agent support

Stream Manager/Media

Server features available

(supervisor, recording)

TDM home agent

1. Call arrives at GW, SIP invite sent to SIP S

2. URS select a TDM Expert agent as target. DN=TDM number

3. GW ip@ in Contact field of DN. SIP S contacts the GW to have the number dialed

4. Call is sent to the TDM phone

5. GW let SIP S know if (i) connected, (ii) busy, (iii) ringing no answer

6. If call answered, Screen pop delivered

Stream Manager/

Media

Server

SIP S

VoIP Gateway

PSTN


Service Provider – Managed Services

POP

Media

Gateway

Customers

SIP Value Prop:

Ease of Deployment

Scalability

GCIP Control of

Business Logic

Light Footprint

Company #1

GCIP

Company #2 Company #3

GVP

SIP

Server

PSTN

GCIP GCIP

IP Network


Virtualization

IP Gateway

Legacy PBX

Agent

Router (URS)

SIP Server

(Parking Lot) Media

Server

PSTN

T-Server

SIP Server

(Agents)

Legacy PBX

Agent

Agent

SIP Value Prop:

One solution to cover all

cases

Allows tranparent

migration from one

technology to another

True virtual solution

without IN

What can a customer expect as a Return on Investment?


Return On Investment Criteria

Reduction of voice infrastructure costs Consolidate voice and data networks

Standardization of skills between data and voice towers

Reduction of toll charges Delivery of calls between sites over corporate WAN instead of public

telephone network

Ability to park calls within corporate network

Reduction of “vendor lock” Ability to use commoditized hardware from multiple vendors (phones,

gateways, media servers, IVRs, networks)

Interoperability between vendors

33



True virtualization of contact centers across multiple geographies

Integration between legacy TDM and IP infrastructure

Make use of existing legacy systems without CTI capability

Provide Remote Agents – PSTN and IP

Possibility to reuse IMS core ( just for carriers )

Better resource usage for experts through UC Connect

34



New media available out of the box Video

Call from web, click 2 call etc

IM platforms

Skype

Full integration with Genesys components and solutions Other media types (TDM Voice, Chat, Email, Open Media, etc)

WFM and Reporting

GVP

Third party applications integrated with Genesys

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Specific Features


Enabling Alternate Ringtones

Alternate ringtones can be supported by sending additional information in the SIP Headers

When SIP Server generates a new call via an INVITE, it can add the Alert-Info header, endpoints can then interpret this for custom ringtones

The option sip-alert-info details the ringtone or ringtone location for the endpoint to play. Similar options sip-alert-external and sip-alert-consult apply specifically to

external or consult calls


Media Server Reliability—MSML

SIP Server supports the SIP SUBSCRIBE/NOTIFY method for providing reliable MSML-based media services through Genesys Media Server. By subscribing to the DN used to represent Genesys Media Server for msml-based services, SIP Server is able to determine when a particular instance of Media Server becomes unavailable, and if so, SIP Server can disconnect from the failed media server, then reconnect to an available instance of Media Server, providing continuity for any ongoing msml-based media services.

1. SIP Server sends periodic SUBSCRIBE messages to GVP Resource Manager for each Voice over IP Service DN (service-type is set to msml ) that SIP Server is subscribed to.

2. If a particular instance of Media Server becomes unavailable, Resource Manager sends a NOTIFY message to SIP Server, which lists the status of each media server monitored by Resource Manager. A sample body for the NOTIFY message is as follows:

*msml/<mediaserver-ipaddress>:<mediaserver-port>/out-of-service*

3. SIP Server disconnects all ongoing media services from this Media Server instance, then reconnects with an active Media Server through Resource Manager.

4. All existing treatments and conference connections are re-established on the alternate Media Server instance.


SIP Server - GVP Media Server (Call recording)

SIP Server 8.1.0 is now integrated with Genesys Media Server 8.1.4 and 3rd party recorders.

Integration is done by MSML protocol.

Only Single VOIP DN needs to be configured for MSML Integration.

The Resource management is taken care of by RM.

39


Voice Gateway

SIP Server

Agent’s phone

Call Recording

Mirorred RTPs T-Library

RTPs

SIP SIP SIP

Media Server

SIP Server (Active Recording)

Recording services natively combine with SIP Server

Easy to add options like Screen Capture, QM &

Reporting Services in a 2nd phase

Integration with Genesys Interaction Workspace (Call Recording state & controls)

[GQM Release 8.1.49 & IWS 8.1.200+]

MSR – Media Stream Replication


Target Solutions

Genesys Quality Management - Integration with Zoom Zoom will act as recording server function to provide storage and

management of recorded media, as well as Quality Management

Genesys SIP Server + GVP Resource Manager + GVP Media Server will act as recording client function to provide call control

IP Call Recording Connector - Integration with Nice Nice will act as recording server function

Genesys components (SIP Server +RM+MS) will act as recording client function


Now & Then

New API developers guide Custom integration to 3rd party

desktop applications (free of charge)

Release v8.1.50

Geo - Location for distributed architectures

GVP Recording

Support for Remote/Home Agents

Switch support – ALU, Siemens, Broadsoft

Release v8.1.51

Active Interaction Recording (iWD/Back Office, eMail, Chat)

Skills driven recording

Tighter integration to Router

Future Direction….

Full Multi Tenancy

Further integration to Genesys CIM

Microsoft Lync & Genesys

© 2012, Genesys Telecommunications

Laboratories, Inc. All rights reserved.


What is Microsoft Lync?

Lync client is used for:

IM & presence,

voice and video calls,

desktop sharing & file

transfer,

and ad-hoc conferences.

Lync

Lync Server (formerly OCS) is an Enterprise Real-time Communications Server,

providing the infrastructure for:

Instant messaging

Presence

File transfer

Peer-to-peer and multiparty

voice and video calling

Desktop & Application sharing

And PSTN connectivity,

with a 3rd party gateway or SIP trunk,

through Lync Mediation Server


Genesys and Microsoft Lync

What we already supported:

UC Connect – Contact Center Extension – Experts & Branches Integration

- Presence integration via Genesys UC Connector

- Instant Messaging from Contact Center Agents to Experts & Knowledge Workers

- Business Context on preview & interaction web screen-pop (Lync integrated)

What’s New?

Support for Microsoft Lync with Enterprise Voice

Serving as the telephony platform for Genesys based Contact Center!

- Genesys announced SIP Server integration with Microsoft Lync Voice in June 2012.

- Integration has been certified by TekVizion labs to the Microsoft Lync Contact Center Interoperability Specification.

- Announced on Microsoft TechNet as a Certified Contact Center Application

Lync


How do we integrate with Lync platform? Lync

1) SIP Server – Voice via Lync Mediation Server 2) UC Connector - Presence integration (server side) - On Hook/Off Hook, Busy, Out of Service/Back In Service

3) Plug-In for Windows Lync client To answer and to initiate Enterprise Voice calls through Genesys Interaction Workspace


How does that work? Lync

SIP signaling bridged

via SIP Server

Customer initiates a call

to the Contact Center

Caller is connected to Media Server

for Qualification & Music

Queuing the call

With Skill & Agent

based routing

When Agent is identified, call is

transferred via Mediation Server

Interaction Workspace toolbar Can be expanded or minimized

•Interaction pop-up window with case data and actions

•3rd Party Call Control on Lync Enterprise Voice call

Customer’s call is connected to the

Windows Lync client

Interaction toast with case data and ability to

Accept/Reject


What can we provide & support with Lync?

Inbound SIP (Voice)

- Advanced Routing (Skills, Agents, Statistics, Business Rules, Context Services)

- Agent’s call control (Make Call, Hold/Retrieve, Transfer, Consult, Conference, …)

- Call Supervision (silent, whisper, barge-in)

Outbound Voice & Proactive Contact

- For Callback/Surveys/Marketing/Collections/….

Genesys Voice Platform

- For Self & Assisted Service (VXML, TTS, ASR, …)

Genesys Quality Management

- Call recording for Compliancy and Performance Management

Operational Reporting & Insights

- Real-Time & Historical

Along with: eServices, iWD, Workforce Management, Social & Mobile Engagement,, …

Lync

And UC Connect Contact Center Extension Leveraging Microsoft Lync

For Experts & Branches integration

GQM


Phase 1 Environment Requirements

Supported endpoints • Lync soft client

• Lync USB phones

Supported Agent Desktop client: • Genesys Interaction Workspace

The following requirements apply • All call control must be performed using 3rd party call control from

Interaction Workspace. In other words, the Lync client cannot be used for hold/transfer/conference/supervision of Genesys calls

• In Phase 1, presence status mapping is performed from Lync to Genesys (one way). Extension to bi-directional presence mapping is planned for next phase.

• Independent hard phones are not supported

Lync


UC Connector - Modes of operation with Lync

Lync voice integration: presence gateway mode

• Used only in the Contact Center in integration with Lync Voice

• Used to pass presence information between Lync and Genesys

• No UI: the UC Connector subscribes to Lync presence to get back to StatServer

• No licensing requirements: part of the SIP Integration license and package

• Users are not aware of the UC Connector - it is simply a component of the system

Enabling customer service for Knowledge Workers

• UC Connect package bundles CIM, SIP Server / T-Servers, and UC Connector

• Separate license needed

• Implements special routing / capabilities for Enterprise Knowledge Workers

• Provides Interaction Preview and 3PCC pop-up windows

• Also subscribes to Lync (or other UC system) presence to relay status to Genesys

Lync


Genesys with Microsoft Lync Voice

Is officially certified and supported by both parties - Microsoft and Genesys

Fully integrates with all the Genesys portfolio (iWD, eServices, GQM, WFM, ….)

Can be part of an heterogeneous voice environment (multi-site support for PBX with CTI, SIP & Lync)

Roadmap Details

Date 2013

Features

• Support for two-way presence propagation (adding Genesys to Lync mapping) • Support for “any” agent desktop client • Support for Routing Video Calls • Support for 1PCC - using Lync Client for call control (hold/transfer/conference) • Full Support for Standalone Phones like Polycom CX500/600/700 • Lync Contact List displayed in Interaction Workspace

Lync


Reporting Ready

Available in Stand Alone or as a Contact Center Extension

Zero footprint on Knowledge Workers’ desktop

UC Connect for the Knowledge Worker UCC

Telephone rings, and pop-up adds

call control commands

Single point of Presence for all

business applications

Combined with agent capacity, determine status

for routing and reporting

Ensure availability with broadcast & previews

Business Context Media Integration & 3PCC Control

Knowledge worker sees

context and clicks on

“Accept”


UC Connect – Current & Future

UC Connect is also available with:

- IBM Lotus Sametime 8.5,

- Microsoft OCS 2007 & Microsoft Lync

- And as Stand Alone Web client!

With Microsoft Lync, UC Connect can also work with Enterprise Voice

- Knowledge Workers & Experts can use Lync client as phone and toolbar.

Roadmap IBM Lotus SameTime 8.5

Microsoft OCS 2007 & Lync

Web Stand Alone

Interaction preview reporting events

Q3-2012 Q3-2012 Q3-2012

Explicit login / logout Q3-2012 Q3-2012 Q3-2012

iWD integration Q1-2013 Q1-2013 Q1-2013

Launch interaction from KW desktop

Q1-2013 Q1-2013 Q1-2013

Active-active HA mode Q1-2013 Q1-2013 Q1-2013

UCC

SIP Voicemail


• Fully Integrated with Genesys SIP

• Agent Voicemail

• Group Voicemail

• Message Waiting

• User Web Portal

• 11 Languages

• High Availability

• Up to 15,000 mailboxes on Off The Shelf Hardware

• Red Hat Enterprise Linux 5.x and Windows 2008

• Simplified All-in-one Installation

Genesys SIP Voicemail v8.1.1 - Features


Functions

Administrative Configuration Manager or Genesys Administrator

- Create mailboxes

- Finalize a mailbox deletion (starts with web GUI)

Web GUI

- Delete messages

- Lock or unlock a mailbox

- Reset the password

- Set/change maximum number of messages

User - Leave messages using local phone or a remote telephone

- Access GSVM via phone or web browser

- Change password

- Manage greetings – extended absence, personal and alternate


Architecture – Standalone Deployment

http://docs.genesyslab.com/wiki/index.php?title=File:Voicemail_architecture_811.png


Architecture – High Available Deployment

http://docs.genesyslab.com/wiki/index.php?title=File:Solution_HA_arch.png


Determine when calls are directed to Voicemail

SIP Server needs to know when to send an interaction to Voicemail.

Methods include: Using No Answer Supervision options for SIP Server

Using one or more Dial Plans to direct calls

Via a routing strategy

Using call forwarding options on agent’s endpoint (if available)


Message Waiting Indication (MWI)

• Sip Server supports MWI to SIP endpoints and Tlib applications

• Sip Server supports “implicit” subscription - NOTIFY is sent to any registered SIP endpoint even if no explicit subscription for

MWI has been done (configurable)

• SIP NOTIFY uses standard “message-summary” event package.

• A mailbox can be shared between several extensions, agent logins or both


Supported Environment – 8.1.1 and higher

Platforms

• MS Windows 2008 (64-bit)

• Red Hat Linux 5+ (64-bit)

Languages

• German (Standard)

• English (UK)

• English (US)

• Spanish (Spain)

• Spanish (Latin America)

• French

• Italian

• Japanese

• Portuguese (Brazil)

• Russian

• Chinese (PRC) o Change default language using ‘locale’ option

o Change individual mailbox language from settings dialog box in the web interface.


Genesys SIP Voicemail Capacity & Hardware Requirements

Hardware Capacity

1 x Quad-Core CPU Example: Intel Xeon E5405 2+ GHz 8GB RAM 80GB storage for messages

5,000 Mailboxes

100 Voice Connections

2 x Quad-Core CPU Example: Intel Xeon E5405 2+ GHz 16GB RAM 240GB storage for messages

15,000 Mailboxes

150 Voice Connections

2 Servers required for High Availability

VMWare ESXi 4.1 & ESXi 5.0 Supported


Administration

Genesys Administrator for Voicemail Server configuration

Voicemail Admin Portal for Admin Control of Individual Mailboxes

(HTML 5 browser required)

Genesys Administrator for enabling voicemail

for Agent or DN (by specifying mailbox)


Web GUI - Administrator

Access via: http://[voicemail server]:8080/voicemail-web/admin


Voicemail Retrieval

Option 1:

SIP Phone

Option 2:

Remote Phone

Option 3:

Web Portal


Web GUI - User

Access via: http://[voicemail server]:8080/voicemail-web/browse


Message Waiting Indicator (MWI)

Each agent or DN is associated with a single mailbox

MWI for this configured mailbox is sent to the SIP phone of the agent/DN

MWI for this configured mailbox is also sent to agent desktop app (may require minor customization to show MWI)


SIP Server Voicemail Options

SIP Server now includes a new option, info-pass-through, that lets you control whether SIP Server will pass SIP INFO messages to a remote device. Configure this option on the destination DN. For example, if DN1 and DN2 are connected, and DN1 sends an INFO message to DN2, SIP Server analyzes the info-pass-through option on DN2 to decide if the INFO message should be passed through or filtered out.


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Configuration – Agent SIPServer

Physical GSVM DN declared as “Voice Over IP Service” Name = contact in GSVM trunk to Agent SIPServer.

“contact” = URI of GSVM SIPServer host:port.

“service-type” = voicemail

One for each GSVM.

Define the mailbox of DNs or Agent Login: Option gvm_mailbox any string (digits only).

Must be unique, but can be assigned to multiple DNs.

SIP Server Business Continuity


Features

• Two site deployment. Active / active operation

• Near real-time failover

• Automatic failover

• Complete Genesys solution

• Database redundancy

• Extended operation of surviving site, with full HA

• Single configuration of common data

• Scalability


Features

• Two site architecture. The sites applications share the same Configuration Server.

• Redundant databases - Configuration database replicated between sites

- Other databases maintained independently at each site

• Peered SIP Server - SIP Servers are united in DR-pair

- SIP Switches use synchronized configuration for some types of DNs and agent logins

- Agent SIP phones register with both SIP Servers. It requires dual registration phone capability.

- Agent desktops can log into SIP Server at either site, but have a preferred site

- SIP Servers coordinate to deliver routed calls, internal calls, external calls from either SIP Server to agents via the SIP Server at which they are currently logged in

- 2 HA pairs of SIP Servers in active / active configuration

Agents can originate calls via either SIP Server


Architecture Overview – Solution Components

1. Solution layers (MGW / Data Center / Contact Center) may or may not be co-located.

2. Each solution layer can be scaled horizontally, independently of other layers.

3. Stacked components indicate local HA redundancy. Can be active / standby, load balanced or parallel depending on the component.

MGW MGW MGW

SIP-S SIP-S

URS

StatSvr

SCS

MS

MFW

CSP

RM

ICON MCP

Media Svr DBMS

LM MS

DBS

ICON GIM

GI2

Reporting

StatSvr

NSS

OCS

Sup / Admin Agent Sup / Admin Agent

SIP-S SIP-S

URS

StatSvr

SCS

MFW

CSP

RM

ICON MCP

Media Svr DBMS

LM MS

DBS

ICON GIM

GI2

Reporting

StatSvr

NSS

OCS

Media Gateways

Data Centers

PSTN

Contact Centers

Admin Admin

CS CS

Site 1 Site 2

IW IW


Architecture Overview – Framework, DBMS, Licensing

1. Active Config Server at Site 1, redundant Config Server at Site 2

2. Config Server Proxy on each side provides survivable local copy of CS DB.

3. Distributed SCS communicating via Message Server.

4. Configuration Database replication (e.g. Oracle Golden Gate) from primary DBMS to backup DBMS.

5. License Manager on each site provides technical licenses for local applications. Each site must have sufficient licenses to support DR scenarios.

MGW MGW MGW

SIP-S SIP-S

URS

StatSvr

SCS

MS

MFW

CSP

RM

ICON MCP

Media Svr

CSDB

LM MS

DBS

ICON GIM

GI2

Reporting

StatSvr

NSS

OCS

Config DB Replication

Active Active


SIP-S SIP-S

URS

StatSvr

SCS

MFW

CSP

RM

ICON MCP

Media Svr

LM MS

DBS

ICON GIM

GI2

Reporting

StatSvr

NSS

OCS

Media Gateways

Data Centers

PSTN

Contact Centers

1

2

3

4

5 5

Admin Admin

Standby

Site 1 Site 2

CSDB CS CS

1

IW IW


Architecture Overview – Administration

1. Users normally connect to Genesys Administrator at “Desktop Preferred Site.”

2. If preferred site fails, users connect to Administrator at other site.

MGW MGW MGW

SIP-S SIP-S

URS

StatSvr

SCS

MS

MFW

CSP

RM

ICON MCP

Media Svr

LM MS

DBS

ICON GIM

GI2

Reporting

StatSvr

NSS

OCS


SIP-S SIP-S

URS

StatSvr

SCS

MFW

CSP

RM

ICON MCP

Media Svr

LM MS

DBS

ICON GIM

GI2

Reporting

StatSvr

NSS

OCS

Media Gateways

Data Centers

PSTN

Contact Centers

Admin Admin

CSDB CSDB

CS CS

Site 1 Site 2

IW IW


Architecture Overview – Peered SIP Servers

1. Agent SIP endpoints have SIP registration on the both SIP Servers simultaneously.

2. Agent desktop normally logs into “Preferred Site” SIP Server, but logs into other site if preferred site is not reachable, or rejects login.

3. Peered SIP Servers check mutual availability and coordinate to deliver routed calls, internal calls and external calls to each agent via the SIP Server at which agent is logged in.

4. Agent desktop can connect to Config Server Proxy at either site, but gives preference to Desktop Preferred Site.

5. Current IW implementation utilizes simultaneous T-Lib connection to the both SIP Servers. However, Agent logged in on one site only.

MGW MGW MGW

SIP-S SIP-S

URS

StatSvr

SCS

MS

MFW

CSP

RM

ICON MCP

Media Svr

LM MS

DBS

ICON GIM

GI2

Reporting

StatSvr

NSS

OCS


SIP-S SIP-S

URS

StatSvr

SCS

MFW

CSP

RM

ICON MCP

Media Svr

LM MS

DBS

ICON GIM

GI2

Reporting

StatSvr

NSS

OCS

Media Gateways

Data Centers

PSTN

Contact Centers

1

2 2 4 4

Admin Admin

3

1 1

1

CS CS

Site 1 Site 2

IW IW


Architecture Overview – Reporting

1. Redundant GIM instances at both sites operate in parallel, fed by ICON DBs at both sites.

2. ICON data that has not been transferred to GIM at time of site failure will be lost.

3. Users can retrieve historical reports from GI2 at either site.

4. Real-time reports: CC Pulse connects to Stat Servers at either site.

5. Every Stat Server monitors both SIP Servers.

MGW MGW MGW

SIP-S SIP-S

URS

StatSvr

SCS

MS

MFW

CSP

RM

ICON MCP

Media Svr

LM MS

DBS

ICON GIM

GI2 StatSvr

NSS

OCS

IDB

IMDB


SIP-S SIP-S

URS

StatSvr

SCS

MFW

CSP

RM

ICON MCP

Media Svr

LM MS

DBS

ICON GIM

GI2 StatSvr

NSS

OCS

Media Gateways

Data Centers

PSTN

Contact Centers

1

3 3

5

4

Admin Admin

IMDB

IDB 1

CS CS

Site 1 Site 2

IW IW

4

5


Call Delivery – Premise Call Routing

.

1. A call arrives a SIP Server on Site 1.

2. SIP Server requests routing instructions from URS.

3. URS selects the best agent to handle the call. In this example, the selected agent is logged in on the other SIP Server. URS requests SIP Server to route the call to the agent. Agent Reservation must be enabled.

4. URS passes Site 2 in the location, so SIP Server transfers the call to the Site 2 SIP Server using ISCC.

5. Call is delivered to agent.

Media gateways distribute incoming traffic across both sites.

Site 1 Stat Server has visibility to agents logged into both SIP Servers

SIP-S SIP-S

URS

StatSvr

SCS

MFW

CSP

RM

ICON MCP

Media Svr

LM MS

DBS

ICON GIM

GI2

Reporting

StatSvr

OCS

MGW MGW MGW

SIP-S SIP-S

URS

StatSvr

SCS

MS

MFW

CSP

RM

ICON MCP

Media Svr

LM MS

DBS

ICON GIM

GI2

Reporting

StatSvr

OCS

Site 1 Site 2


Media Gateways

Data Centers

PSTN

Contact Centers

Admin Admin

1

3

5

4

CS CS

IW IW

2


Call Delivery - DR-pair Target detection & Forwarding

• The Party that originates the direct inbound (or internal call) call does not know the current agent’s location. So call can arrive to any server in a DR pair.

• “Target Detection & Forwarding” procedure takes measures to deliver the call to the proper site in Business Continuity deployment.

- The procedure is applied to all calls to be delivered to the agent DNs.

- If destination is ready for call, call is delivered to it. Otherwise, call is forwarded to the DR-peer.

- This activity related to agent availability only. There is no intention to try the second site destination if phone is inaccessible on the first site.

- To avoid the infinitive loop of forwarding procedures in case of logged out agent, the forward SIP messaging includes private URI parameter in the ‘Contact’ (for 302) or ‘Refer-To’ (for ‘REFER’).

Contact: <sip:7101@dr-peer:5060;x-genesys-dr-forward>

- So, if call has been already forwarded from peer site then current SIP Server must try to deliver it to DN requested independently of agent status.


Failure detection

• Sip Server - Utilizes the dedicated trunk which points to its peer.

- Trunk name is defined in application option “dr-peer-trunk”.

- ”The ‘Out of Service’ check mechanism configured on this trunk provides the failure detection functionality( options “oos-check” and “oos force”.

• Desktop (T-Lib client) - Desktop must use the ADDP protocol to detect site availability.

SIP end point -failure detection depends on the phone model - Bria: Supports “Send SIP keep-alive” option. End point utilizes empty

messages sent to SIP port and relies on ICMP errors to detect

- Polycom SoundPoint: it does not have keep –alive mechanism. The failure could be detected by failing of outbound call or renewal of the registration.

- ALU 40xx uses OPTION request to check the SIP Server availability. Important that it does not continue sending OPTIONS to a dead site. So, it cannot detect the primary peer recovering. As result, it continues to work with secondary peer until failover on secondary site.


Graceful migration

• BC supports graceful migration of operations from two active/active SIP Server Peer sites to a single site

• Graceful migration includes administrative and automatic parts: 1. Configure your MGW to stop sending calls to the SIP Server Peer site that you intend to

shutdown.

2. Possible step is starting the dedicated strategies on URS.

3. Using Genesys Administrator, then initiate a graceful shutdown of the SIP Server

• During the graceful shutdown procedure SIP Server 1. Moves all agents into NotReady state

2. Logs out all agents who does not have active calls (with appropriate reason code ) and declares DN Out of Service

3. Rejects all INVITE with configurable final response (option “shutdown-sip-reject-code”)

4. Forbids any call created T-Request .

5. Rejects all agent login or ready requests.

SIP Server waits until last call is finished and then stops.


Supported Environment

Soft switches Siemens OpenScape

Voice

Agent Desktop Interaction Workspace

(8.1.1+)

SDK support for Interaction Workspace and 3rd party desktops (8.1.0+)

Platforms

• MS Windows 2008

• Red Hat Linux 5

• AIX

• Solaris

Database

• Oracle

Endpoint Types

• CounterPath Bria

• Polycom Soundpoint IP

• ALU 4000

• SIP Endpoint SDK

• Remote agents (e.g. TDM)


Administrator in BC deployment

• Genesys Administrator takes care of synchronization in business continuity environment.

- Tool to create a new switch object or select an existing switch object to be synchronized with the original.

- Monitors updates of Persons and Places related to DR-pair of SIP Servers and synchronizes the changes on the fly.

- Tool to check the synchronization consistency of CME objects related to DR-Pair of SIP Servers


1. Synchronize the DR pair of switches

Sync the DR pair of switches using GA ‘Sync Switch’ wizard.


1. Synchronize the DR pair of switches

The Wizard sets up the switches as peers, synchronizes switch-related elements between them, and then keeps them synchronized.


Interaction Workspace support of BC deployments

• Configuration supports information about DR-pair and preferred site.

• Supports simultaneous dual SIP registration of SIP end points

• Can recognize Reason codes for forced logout and properly react on it.

• Supports site failure detection and switchover mechanism.

Whats New in SIP Server 8...Integration with Softswitch supported (Siemens, Cisco & others) Traffic...

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