Configuring Basic Enterprise

Voice Functionality
Voice Routing
msRTCSIP-Line
msRTCSIP-PrivateLine
msRTCSIP-PrivateLine
SIP URI
1

PSTN
PSTN Fallback
Fallback for
for
CAC
CAC and Inbound
Inbound Routing
and Network
Network Routing
Dial Plan 7 Reverse
Reverse Number
Number Lookup
Lookup Outages
User=phone Match
Emergen
Emergen Yes No match
External Normalization
Rules
cy
cy 8 Apply
Apply Called
Called
Call?
Call? Party Prefs
EA
EA Select usages
P?
P? 2 Usages on UC
UC Endpoint
Endpoint Receives
Receives
Internal
Normalization Rules
No 9 From
From trunk
trunk
with inbound Call
Call
with
usages? trunk
usages? Usage from
Emergency
Emergency location
Client-side Yes call?
call?
Global? policy
normalization
normalization 3 Global?
Conferenc
Conferenc
Usages of 10
e dial-out?
dial-out? meeting
e
organizer
RFC
RFC 3966
3966 No Unassigne
Unassigne
d number?
number? Announcement
Starts with + d Announcement or or
Call Park
Call Park Application
Application
Call
Call Park?
Park?
Usages
Location-
Location-
Gateway Mediation from
based
based
Server routing? network
Dial Plan routing?
Referred-
Referred- site
by?
Usages of
by?
referrer Convert
Normalization Rule Convert ##
4 to Local Format
Normalization Rule
Normalization Rule
Usages of 13
caller
Trunk
Trunk Configuration
Configuration //
Number
Number Translation
Translation
Must
Must Match
Match Selected Routes
6 A Rule Call Park Orbit Range policy/usages
PSTN Usage
12 Route
Gateway
Gateway // IP-PBX
IP-PBX // SIP
SIP Trunk
Trunk
5 PSTN Usage
Route
Route
11 PSTN Usage Route External
External Endpoint
Endpoint Receives
Receives
Call
Call
14
Dialing Routing &
Behaviors Authorization
Number Normalization and E.164

Country Code National Destination Subscriber

Code (Optional) number
National (significant) number
1 to 3 digits Maximum = 15 cc = 12 to 14 digits
31 (Netherlands) 20 (Amsterdam) 500 1500

1 (US) 425 (Washington) 882 8080

Country Code Group Subscriber number

Identification
Code
3 digits 1 digit Max = 15 (cc + gic)
= 11 digits
599 (Netherlands 7 (Bonaire) 500 1500
Antilles)
Scoping Configuration Items and Policies

Global Contoso

Site
Chicago London

Dublin-
Pool
Chicago-1 Chicago-2 1

User
 Dial Plans

A set of normalization rules that translate dial strings to full, unique

numbers
^ match the start $ match the end
Normalization
rules are \d match any digit \d* 0 or more digits
specified using
\d{5} any 5 digits [135] 1, 3, or 5
regular
expressions (13)|(17) 13 or 17 [1-5] 1 through 5

() captures the enclosed characters for referring to them in the

result as $1, $2, $3, etc.
Normalization and Regular Expressions

Dial plans perform normalization by using

regular expressions
Skype for Business Control
Panel
Or built from scratch by using
standardized regular
expressions
Example Normalization Patterns

National dialing
^([2-9]\d\d[2-9]\d{6})$ +1$1 (NANP)
^0(\d{10}) +44$1 (UK)
Include national and international dialing
prefixes
^011(\d*) +$1
Extension range (e.g. 15xx-35xx)
^((1[5-9]|2[0-9]|3[0-5])\d{2})$
+1206555$1
 Address book normalization

No more text file.

Yay!
Not handled
during in-place
upgrade
Cmdlet for
importing existing address book rules
Import-CsCompanyPhoneNormalizationRules
-Filename "Company_Phone_Number_Normalization_Rules.txt" -Identity Global
Overview of Routing and Authorization

Voice Policies PSTN Usages Routes

User authorization Purpose (usage, Called number

Class of service callers intent) Cost of call
Voice feature set Calling location
Priority
Route Planning

International

National Premium

National

Local

Internal

Routes for the gateways in Munich

DE Internal ^\+49895550[12]

DE Munich Local ^\+4989

DE Germany ^\+49

DE Europe ^\+(49)|(31)|(33)|(32)|(34)|(351)|

DE International ^\+
Voice Policies
Can be assigned per user, site, or global.
Can be by PS:
New-CsVoiceRoutingPolicy Identity <PolicyID> -Name
<PolicyName> -PstnUsages <Usage1>, <Usage2>
Not only for users. Also useful to address Common
Area Device requirements:
Assign a Voice Policy to a common area phone to prevent
misuse and high cost.
Provides admins with flexibility to control user
voice entitlements:
PSTN Usage

Control dialing capabilities (Class of Service) by

assigning PSTN usages
A Public Switched Telephone Network (PSTN)
usage record specifies a class of call (such as
internal, local, or long distance) that can be
made by various users or groups of users in an
organization
By themselves, PSTN usage records do not do
anything. For them to work, they must be
associated with the following:
Voice policies, which are assigned to users
Routes, which are assigned to phone numbers
Call Forwarding and Simultaneous Ring

Lync Server 2010: An administrator can enable or

disable call forwarding and simultaneous ring
through the user voice policy
Lync Server 2013 & Skype for Business: Enables
call controls to introduce a flexible call-
authorization mechanism for forwarding and
simultaneous ring calls
By using this feature, a company can restrict
calls forwarded by users or through simultaneous
ring
Local numbers only, to aid in cost control
Internal Skype for Business users only, for security
policies
Any custom authorization rule set up by the
Call Forwarding and Simultaneous Ring

PSTN Usage
Voice Policy Local,
Redmond International
Call PSTN
Usages
Call
Forwarding Internal
Skype for
Business
users only
Simultaneous
14
Ring Custom
PSTN
Usages

PSTN Usages
Custom
Usage
Voice Routes

A voice route associates destination phone

numbers with one or more public switched
telephone network (PSTN) gateways or SIP
trunks, and one or more PSTN usage
records
A route is selected based on a matching
pattern
PSTN usages control if a user is allowed
to use the route
Routes are associated with one or more
trunks defined in Topology builder
Trunk Configuration

Allow for centrally managing number

formatting prior to routing to PBX/PSTN for
both the calling and the called number
Assigning DIDs to a User

DID is a term used by the

telecommunications industry and stands for
Direct Inward Dial (DID):
DID numbers are globally unique
DID ranges/blocks are acquired from the
telecom provider
DID numbers enable external users to
connect to a Skype for Business user
directly
Are assigned to a user when enabling for
Enterprise Voice
Specifying a Line URI

DID numbers can be defined in two formats:

tel:+14258828080;ext=8080
tel:+14258828080
Recommendation: Specify extensions (ext=)
for all users to:
Optimize PIN authorization for devices and dial-in
conferencing
May need to deal with ;ext= in trunk
normalization for cases of PSTN reroute
Internal-OnlyUsers Without DID

The full URI points to the switchboard or Exchange

AA number:
Users will have a unique ext=xxxx
Example:
User A tel:+14258828080;ext=51855
User B tel:+14258828080;ext=51856
User C tel:+14258828080;ext=51857
Base number should point to Exchange AA with
number:
tel:+14258828080;ext=1
Normalization of the inbound number should add
;ext=1 so that the unique number of AA can be
found by using a reverse number lookup:
^(\+14258828080)$ $1;ext=1
Dial Plan Design Approach

Record all existing dialing habits

Consider the current dial plan
Understand the Gateway and Mediation
server locations
Understand the customer requirements
Dial Plan Design Approach

Migration strategy
Skype for Business or PBX Phone vs. Skype for
Business and PBX phone?
Keep the existing DIDs or get new numbers when
migrated to Skype for Business?
Implement changes or copy the existing numbering
plan?
Define the routes
Define user voice policies (Classes of Service)
Incorporate requirements for least-cost routing and
PSTN rerouting and fallback, if needed
Use PSTN usages to link appropriate routes to
the needed voice policies
Real World Scenarios and
Recommendations
Copying existing dialing habits is not always
a good idea
Some are just there to accommodate the
PBX
Examples of unnecessary dialing rules:
Scenario Remark Recommendation
Dial a 9 to seize an PBX-specific Avoid creating
outside line behavior normalization rules for
Example: 9 this habit
0031205001500

Dial 00 for an Country-specific Create a normalization

international number behavior rule that translates to
Example: 00 Not required, but E.164
31205001500 everybody uses it

Prefix an internal PBX-specific Avoid creating

number with a PBX- behavior normalization rules for
 Route PlanningA Real World Example

Example routes for the gateways in Europe

Germany (DE) Munich
DE Internal ^\+49895550[12]
DE Munich Local ^\+4989 International
Europe DE Germany ^\+49 National Premium
GatewayDE Europe ^\+(49)|(31)|(33)|(32)|(34)| National
DE International ^\+ Local
Internal

Asia USA
International
Gateway Gateway
National Premium
National International
Local National Premium
Internal National
Local
Internal
Number Blocking

Traditional Method

Alternative Method
Microsoft Official

Course
Voice Applications
Call Park - Features

Call Park and Retrieve

Orbit (number) returned when call is parked
Parked user is listening to Music on Hold (MoH)
Call can be retrieved from PBX phone dialing orbit
Safe-retrieve: only retrieve my parked call
Ringback
Calls not retrieved are transferred to person who parked
the call (after timeout)

Transfer to fallback destination

Calls not retrieved and ringback failed are forwarded to
 Skype for Business Call Parking

A call can be parked if the user is enabled

for Call Park functionality

An available orbit is
automatically offered to the user parking
the call
Skype for Business Call Retrieval

Dial the orbit like any

other extension

Click Retrieve button (performs a safe

retrieve) or copy the link into an IM
message
Unique ID to identify the call

Parker receives notification of who retrieved

the call
Call Park Ringback

After pre-configured timeout

(CallPickupTimeoutThreshold )
Call rings back
User can click the Answer
the Call button
Call can be ignored
Call cannot be redirected
Call is not forwarded to voice mail
Deploying Call Park Services

Call Park services are installed when a

server is enabled for Enterprise Voice

Enable Call Park for the end-user in the

Voice Policy (disabled by default)
Defining Call Park Ranges

Configure orbit range and destination

pool (global scope)
Orbit Range should be globally unique
May not include DID numbers Se01.adatum.local

Ranges can be configured in Skype for Business

Control Panel
Must start with # or *, or 1-9.
0 is not allowed as a starting character
Must be the same length (max. 9 characters)
Should not exceed 10,000 orbits per range
Should not exceed 50,000 orbits per pool
Exclude Call Park orbits from Normalization
Option to use #100 to #200
A single pool can have multiple orbits
Call Park Management

Optional settings can be changed, as

follows:
Music on Hold can be changed or disabled
(service scope)
Ringback attempts (1-10) (site/global scope)
Ringback timeout (10-600s) (site/global scope)
Fallback destination (site/global scope)

All configuration through PowerShell, except

Orbit range
New-CsCpsConfiguration
New-CsCpsConfiguration -Identity
-Identity site:<sitename
site:<sitename to
to apply
apply
settings>
settings>
[-CallPickupTimeoutThreshold
[-CallPickupTimeoutThreshold <hh:mm:ss>]
<hh:mm:ss>] -[EnableMusicOnHold
-[EnableMusicOnHold
<$true
<$true || $false>]
$false>]
[-MaxCallPickupAttempts
[-MaxCallPickupAttempts <number
<number of
of rings>]
rings>]
[-OnTimeoutURI
[-OnTimeoutURI sip:<sip
sip:<sip URI
URI for
for routing
routing unanswered
unanswered call>]
call>]
Call Park Deployment Process

create the orbit ranges in the call park orbit table

New-CSCallParkOrbit
New-CSCallParkOrbit and associate them with the Application service
that hosts the Call Park application

Set-CsCpsConfiguration
Set-CsCpsConfiguration Use the cmdlet to configure Call Park settings

Set-
Set-
CsCallParkServiceMusic
CsCallParkServiceMusic Optionally, customize the music on hold
OnHoldFile
OnHoldFile

Set-CSVoicePolicy
Set-CSVoicePolicy Configure voice policy to enable Call Park for users
 Park and Retrieve Call Flow

Step 1:
Alice calls Bob, incoming call

who is using
Skype for
Business Server Front End User Bob
2015

incoming call

Caller Alice Mediation Server

 Park and Retrieve Call Flow (2 of 7)

Step 2:
Alice is now connected to
Bob
Media flows from Alice to
Bob
User Bob

ow
Front End

Fl
ia
ed
M
Media Flow

Caller Alice Mediation Server

 Park and Retrieve Call Flow (3 of 7)

Step 3:
Alice wants to speak to Park Call
Charlie
Bob issues a call park User Bob
command to the Call Park Front End
Service, requesting an orbit

ow
Fl
ia
ed
M
Media Flow

Caller Alice
Mediation Server
 Park and Retrieve Call Flow (4 of 7)

Step 4: Orbit
123
Alice is put on hold,
receiving Music on Hold
from the Call Park Service Front End
Bob receives a Call Park User Bob
orbit
Media
Flow

Media Flow

Caller Alice Mediation Server

 Park and Retrieve Call Flow (5 of 7)

Step 5:
Bob shares the Call Park
orbit with Charlie through
an internal paging
system, IM, or some Front End User Bob
alternate method
Media Orbit
Flow 123
(paging)

Media Flow

Caller Alice Mediation Server

User
 Park and Retrieve Call Flow (6 of 7)

Step 6:
Charlie dials the orbit
number in an attempt to
retrieve the parked call
Front End

Re
tri
ev
Media

e
12
Flow

3
Media Flow

Caller Alice
Mediation Server
User
 Park and Retrieve Call Flow (7 of 7)

Step 7:
Alice is now directly
connected to Charlie

Front End

Media Flow Media Flow

Mediation Server User

Caller Alice Charlie
Purpose of the Unassigned Number
Feature
Handles incoming calls to numbers valid to
the organization but not assigned to users
or (desk) phones
Avoids busy tones or error messages if the
user misdials
Incoming calls can be transferred to
predetermined:
Phone Numbers
SIP URIs
Voice Mail
Announcement service
Announcement Service

Create an Announcement through Windows

PowerShell
New-CsAnnouncement
New-CsAnnouncement -Identity
-Identity
ApplicationServer:se01.tailspin.local
ApplicationServer:se01.tailspin.local -Name
-Name "Number
"Number Does
Does Not
Not
Exist"
Exist"
-TextToSpeechPrompt
-TextToSpeechPrompt "Welcome
"Welcome to
to Tailspin,
Tailspin, the
the number
number you
you dialed
dialed
does
does not
not exist.
exist. You
You will
will be
be forwarded
forwarded to
to the
the operator"
operator" -Language
-Language
"en-US"
"en-US" -TargetUri
-TargetUri "sip:brad@tailspin.com
"sip:brad@tailspin.com

TextToSpeechPromptA text-to-speech (TTS) prompt

TargetURIThe Uniform Resource Identifier (URI) to which
the caller will be transferred after the announcement has
been played
At least one Announcement should exist before you
can create a number range
Deploying the Unassigned Number Feature

Create an unassigned number range from the

Skype for Business Control Panel
Range may overlap with existing DID; numbers in use
automatically excluded
Destination server is the end point and plays the
announcement; plan locally, preferably in the same site
Select the previously created Announcement, or
choose to forward the call to an Exchange Auto Attendant

ApplicationServer.se1.adatum.com

Number does Not Exist

 Unassigned Number Call Flow

User Bob
Step 2
Front End

Tr
Ca sfe
an
ll r
Media
Flow
Step 1

Media Flow Media Flow

Step 3 Step 3
Caller Alice Mediation Server User
Charlie
 Unassigned Number Call Flow (1 of 3)

Step
1:Alice has dialed a phone number that
she believes belongs to Bob
The vacant number routing determines Front End
that the dialed number is not a valid User Bob
number
Alice is connected to a special RGS Media
workflow and is notified that the Flow
number is not in use

Media Flow

Caller Alice Mediation Server

 Unassigned Number Call Flow (2 of 3)
Step
2: The special RGS workflow now transfers
Alice to Charlie as configured by the
vacant number announcement
(-TargetURI)
User Bob

Front End

Media Tr Ca
an ll
Flow sf
er

Media Flow

Caller Alice
Mediation Server User
Charlie
 Unassigned Number Call Flow (3 of 3)

Step
3:
Alice is now connected in a voice call to
Charlie

Media Flow Media Flow

Caller Alice Mediation Server User

Charlie
PSTN Conferencing Features

Meeting Features to handle small/mid-size

meetings
DTMF controls
Entry and Exit announcements
Simple join experience
Lobby support for restricted meetings
Unauthorized users wait in the lobby to be admitted
Name recording for unauthenticated users
Integrated seamless with Skype for Business meetings
Scheduling through familiar Skype for Business
interface
Access security by PIN and phone number
authentication
Meeting prompts and guidance in a language of
choice
Meeting Types

Dial-In Conferencing

Reservation-less calls

Managed events
 Meeting Types
Reservation less calls85%
Weekly staff meetings, project meetings and so on
Typically 25 or fewer participants, average of 3-5
attendees per meeting Target for
Majority of attendees are internal Skype for
Business
Frequently contains external attendees
Web attached
Operator-assisted callsLess than 10%
Biweekly/monthly
Roll call, polling, and other large meeting features
From 25-100 attendees
Managed event
Web attached ACPDomain
(Audio
Externally focused calls5% Conferencing
Provider)
With transcription, high touch, max features, and large audiences
100+ participants
Quarterly or less frequent
Web attached Based on Gartner
Study
 User Roles & Permissions
Presenter
Controls meeting
Designated by organizer
Cant designate Federated in advance

Organizer
Implicit role, presenter by definition
If deleted from AD, conferences also removed from
RTC database
Attendee
Everyone who is not a presenter
Cannot add content to meeting
Can only download content if given permissions
Can be promoted / demoted
DTMF Commands

Commands Admin customizable

*1 Automated help Each command can be
configured as * / # + 0-9
*3 Private roll-call Each command can be
*6 Mute/unmute self disabled (unset key mapping)
Exposed through PowerShell
*7 Lock/unlock
(leaders only) End-user discoverable
*4 Toggle silent Shown on the Dial-in
Conferencing webpage
mode (leaders only) Discoverable in conference by
*9 Entry/exit issuing Help command (*1)
announcements
on/off (leaders only)
*8 Open lobby
(leaders only)
 Entry/Exit Announcements

Entry/Exit announcements with names

Announcements are made when participants join and leave
Batching reduces the number of announcements
Anonymous PSTN users are prompted to record their names
Authenticated user names are announced by text-to-speech
(TTS)
Users can skip name recording and join as unknown participants
John
(federated
user) Jane
(PSTN user
anonymous)

MCU
Alice
Authenticated user
Bill
(PSTN user
authenticated)

Simon
Anonymous Skype for Business
2015/Skype for Business Web App
user
Entry/Exit Announcements (2 of 2)

Controlled by
Admin - Entry/exit announcements configuration:
Off Set-CsDialInConferencingConfiguration
Set-CsDialInConferencingConfiguration
Beep -Identity
-Identity site:Redmond
site:Redmond
-EntryExitAnnouncementsType
-EntryExitAnnouncementsType "ToneOnly"
"ToneOnly"
Name, TTS for known users
or Recording for unauthenticated users
Organizer:
Turns announcements on/off at
scheduled time for non-default meetings
Presenter:
Turns announcements on/off during the meeting
Important Settings - Join Experience

Settings related to the join user experience

Default meeting policy (set by
administrator, can be changed by user)
Lobby bypass for PSTN users (set by user)
Deploying PSTN Conferencing Services (1
of 2)
Plan additional Direct Inward Dialing (DID) numbers
and PSTN trunk capacity for (regional) PSTN access
numbers
Consider toll free numbers
Deploy PSTN gateways or configure SIP trunking
Configure access numbers globally or per site:
Assign access numbers to conference regions
Define primary and additional languages (maximum 4)
Configure dial plans with a valid dial-in
conferencing region
Dial-in conferencing regions associate a dial plan with one
or more dial-in access numbers
Deploying PSTN Conferencing Services (2
of 2)
Configure PIN security settings (complexity,
expiration, and so on)
Generate PIN and send welcome email message by
using the PowerShell script
(SetCsPinSendCAWelcomeMail.ps1)
Set-CsClientPin
Set-CsClientPin -Identity
-Identity tailspin\holly"
tailspin\holly" -Pin
-Pin 18723834
18723834

Enable user for PSTN

dial-in (conferencing policy)

Optional
Configure DTMF commands globally or per site
Manage order of access numbers per conference
region (PowerShell cmdlet only)
Managing Conferencing

How many conferences are happening now?

Get-CsWindowsService
The ability to call all Skype for Business services running on
local computer
Audio Conferencing Architecture

Skype for
Skype for Business Front-End Server Business Back-
End Server
Web Components (SQL DB)
Focus Conferencing
(IIS)) Audio Video Database
Conferencing
Join Launcher Server
Focus Factory

Reach Server
IM Conferencing Server

Dial-in
Conferencing
Web Conferencing Server Page
Machine Boundary

App Sharing Process Boundary

Conferencing Server Conference
Announcement
Service Web Application
Conference Auto
Attendant Personal Virtual
Assistant
Audio Conferencing
Group Virtual
Assistant
Multi-Language Support

Caller 1 joins and requests English Voice Applications

Conference Announcement
Service

English
Caller 2 joins and requests English
Group Virtual Assistant
(C1/C2)
Personal Virtual Assistant
(C1)
Personal Virtual Assistant
(C2)

Caller 3 joins and requests German German

Group Virtual Assistant
(C3)
Personal Virtual Assistant
(C3)
Typical PBX deployments

Basic PBX features Add-on ACD Dedicated ACD

(Basic Hunt Group) solution High scale
Fully featured High additional costs
Additional licensing
costs

MoH
Basic hunt groups High scale
Agent sign-in/sign- Busines Superviso High availability
out s hours r Advanced CDRs
Various hunting Basic Live Interoperable with LoB
methods CDRs views applications
Advance
d CDRs

Departmental solutions Internal Large Call Centers

Help
desks,
Small
Call
Centers
Response Group Features

Interactive Voice Response (IVR)

Call queuing
Routing
Agent-side user experience
Infrastructure
Positioning Skype for Business Response
Groups
Basic PBX features Add-on ACD Dedicated ACD
(Basic Hunt Group) solution High scale
Fully featured High additional costs
Additional licensing
costs

Response Group Service High scale

Superviso High availability
Hunt groups and basic r Advanced CDRs
Live Interop with LoB apps
IVRs views
Integration with Skype for Advance
Business presence d CDRs
Agent anonymity
Announcements Internal Large Call Centers
(unassigned numbers) Help
Speech recognition and desks,
Small
TTS Call
Music on Hold Centers
Basic CDRs
Response Group Service (RGS)
enhancements

RGS has been enhanced to improve

scalability
in Skype for Business Server
RGS Agent Group: 800
IVR group: 400
Agents per pool: 2,400
Response Group Management

An Administrator can delegate the

management of response groups to a
Response Group Manager
The Manager role improves the scalability of a
response group deployment by decentralizing the
management of the response groups from the
administrator
The scope of a Response Group Manager is
at a workflow level
A Manager cannot see or modify response groups
for which he or she is not a Manager
Managed and Unmanaged Response
Groups

Administrator(s)

Manager 1 Manager 2
Work Work Work Work
FlowManage Flow
Manage Flow Flow
Unmanage
Unmanage
d d d d

Queue Queue Queue Queue Queue

Agent Agent Agent Agent Agent

Group Group Group Group Group
 Response Group Building Blocks
Agents
Target for incoming calls User 1 answered the last
Enterprise Voice user(s) call
Not a specific RGS object User 2 is the 3rd longest
Member of one or more Groups idle

User 3 is the 2nd longest

Groups idle
Ordered list of agents or
Exchange Distribution Groups User 4 is the longest idle
Routing Method
Attendant Ring1, 2, 3 all at the
same time
Parallel Ring1 and 2 at the same
time (as 3 is in a call)
Longest IdleRing 4, wait 30
seconds, Ring 1, wait 30 seconds,
Membership can be formal or Ring 2, and so on
informal Round RobinRing 2, wait 30
Uses predefined routing methods seconds, Ring 4, wait 30 seconds,
Added to one or more queues Ring 1, and so on
SerialAlways Ring 1, wait 30
seconds, Ring 2, wait 30 seconds,
Formal vs. Informal User Groups

Informal User Group membership

User signs in to the Skype for Business client
User is automatically available as an active agent

Formal User Group membership

User signs in to the Skype for Business client
User must sign in again to
become an active agent
Configuring Queues

Queues
Holds call until agent
pickup
Serviced by one or many
groups
Follows each groups routing
sequence
Various configuration
options
Queue Overflow Action
Queue Timeout Action
Custom Prompts

Target for a Workflow

Configuring Workflows
Sample RGS Scenario - Operator

Classic Operator

Operator with Fallback

Operator with
Fallback
and After-Hours
Service
 Deploying Response Groups

Define agent groups

Skype for Business Control Panel)

Define agent groups

Skype for Business Control Panel)

Define the workflow

(RGS Web Page)
 RGS Call Flow and Agent Anonymity

Ringing RGS Alice calls a Response Group

Call flows differ depending on
Caller Alice
Agent anonymization
Initial call is always targeted at
the Response Group
Establish
ed
RGS Ringing

Caller Alice Agent Bob RGS alerts one or more agents

No agent Agent anonymization:

anonymization:
Agent answers Agent answers
Alice connects directly Alice remains
RGS no longer part of connected through
the call RGS
Agent is hidden
(anonymous)
Establish
ed Establish Establishe
ed
RGS d
Caller Alice Agent Bob Agent Bob
Caller Alice
RGS Call Flow and Agent Anonymity (1 of
4)

Alice calls a Response Group

Call flows differ depending on
Ringing RGS Agent anonymization
Initial call is always targeted at
Caller Alice the Response Group
RGS Call Flow and Agent Anonymity (2 of
4)

Alice calls a Response Group

Call flows differ depending on
Ringing RGS Agent anonymization
Initial call is always targeted at
Caller Alice the Response Group

Establish
ed
RGS Ringing

Caller Alice Agent Bob

RGS alerts one or more agents

 RGS Call Flow and Agent Anonymity (3 of
4)

Ringing RGS Alice calls a Response Group

Caller Alice Call flows differ depending on
Agent anonymization
Initial call is always targeted at
Establish
the Response Group
ed
RGS Ringing

Caller Alice Agent Bob RGS alerts one or more agents

No agent
anonymization:
Agent answers
Alice connects directly
RGS no longer part of
the call

Establish
ed

Caller Alice Agent Bob

 RGS Call Flow and Agent Anonymity (4 of
4)

Ringing RGS Alice calls a Response Group

Call flows differ depending on
Caller Alice
Agent anonymization
Initial call is always targeted at
the Response Group
Establish
ed
RGS Ringing

Caller Alice Agent Bob RGS alerts one or more agents

No agent Agent anonymization:

anonymization:
Agent answers Agent answers
Alice connects directly Alice remains
RGS no longer part of connected through
the call RGS
Agent is hidden
(anonymous)
Establish
ed Establish Establishe
ed
RGS d
Caller Alice Agent Bob Agent Bob
Caller Alice
Group Call Pickup Feature

Added in February 2013 Cumulative Update

for Lync 2013

Allows any user to pickup calls for their

colleagues using their own phones
A user can be a member of only one call pickup
group.

Leverages Call Park application

Similar to, but different from Team Call

Group Call Pickup Feature - Planning

Components Used
Application service
Call Park application
Skype for Business Server Management Shell
SEFAUtil Resource Kit Utility
Clients
Skype for Business, Lync 2013, 2010, Phone Edition
User must be homed on Skype for Business or Lync 2013
Pool with Feb 2013 CU
Users can only be a member of one call pickup
group
DR requires admin to repoint orbits
Group Call Pickup Capacity Planning

Per Front End pool

Per Standard
Metric (with 8 Front End
Edition server
Servers)
Recommended number of
50 50
users per group
Recommended number of
500 60
groups
Maximum number of users
per pool enabled for Group 25,000 3,000
Call Pickup
Maximum rate of incoming
calls to total users enabled
500 60
for Group Call Pickup per
pool per minute
Maximum rate of calls
retrieved by users with
200 25
Group Call Pickup per pool
per minute
Group Call Pickup - Deployment
SEFAUtil (dedicated server)
$Site=Get-CsSite Identity Datacenter1
New-CsTrustedApplicationPool -Identity
"dirsync.Litwareinc.com" -Registrar
"litwarepool.litwareinc.com" -Site $Site.SiteID
New-CsTrustedApplication ApplicationId "sefautil"
TrustedApplicationPoolFqdn server.litwareinc.com" -Port
7000
Enable-CsTopology
Configure Call Pickup Number Ranges
New-CsCallParkOrbit -Identity "Redmond Call Pickup"
-NumberRangeStart *100 -NumberRangeEnd *199
-CallParkService litwarepool.litwareinc.com -Type
GroupPickup
Assign Call Pickup Number to Users
SEFAtuil.exe as@contoso.com /server:pool.contoso.com
/enablegrouppickup:*100
Group Call Pickup Call Flow
Microsoft Official

Course
Configuring and Deploying
Emergency Calling
What Is Location Awareness?

Location Information Service (LIS)

Identifies and populates user location in the
client
Affects routing of emergency calls
Provides street address for E911
"Location Aware" Emergency Routing

Emergency calls can be

routed to specific PSTN
gateways by a location
policy
A location policy can be
assigned to network site
Example: A user roaming
outside North America
A user travelling in Ireland
dials 112
(for emergency)
The call is routed to a local
gateway
Voice Routing
User
User Initiates
Initiates Call
Call
Dial Plan
User phone SIP URI

Normalization Rule
Normalization Rule No No Emergenc
Global
Global
Normalization Rule y
y
?
? Call?
Call?

404: No
Yes
matching Call Park Orbit Range Yes
Dialing rule
Behaviors

Reverse
Reverse Number
Number Lookup
Lookup
Routing and
Authorization No match Match

3. Voice Policy Routes Location Policy

1. Vacant Number
Range
PSTN Usage Route PSTN Usage
Route
2. Call Park Orbit PSTN Usage
Route
Range PSTN Usage Route

Mediation
Mediation Server
Server and
and
Announcement 403: No Trunk
Trunk Configuration
Configuration
Announcement or or
Call route
Call Park
Park Application
Application
found
Gateway/IP-PBX/SIP
Gateway/IP-PBX/SIP Trunk
Trunk Inbound
Inbound Routing
Routing

External Endpoint Receives UC Endpoint Receives

Call
Call Call
Call
E9-1-1 Skype for Business Server 2015
Components

Location Policies
E9-1-1 Configuration

2
Location Discovery

3
2
4
1

Caller
Placing an Emergency Call

3
4a 4b

2
5
1
Security

Caller PSAP
Location Policy Definition

Defines user experience, routing, dial string,

and Security Desk notification
Location Policy Scope

If present, apply the Global Policy

policy assigned to
network site
Note: This is a user
policy that you create and
assign to the network site
Or, if present, apply
the policy assigned
to user
Or, consider the
Topology Site Policy,
and then the Global
Location Policy
PSTN Usage Creation
The location policy PSTN usage selects a
route for the emergency call
This may be the SIP trunk of the service
provider (one usage for all location policies)
or it may be a local gateway (a unique
usage for each policy)
The PSTN usage is not added to existing
Voice policies:
It is used for an emergency call if a user has a
location policy applied
You can create PSTN usage in either Skype
for Business Server Control Panel or Skype
for Business Server
PS>Set-CsPstnUsage
PS>Set-CsPstnUsage Usage Management
Usage Shell
@{add="EmergencyCallsUsage"}
@{add="EmergencyCallsUsage"}
Voice Route Creation

The Voice Route allows calls to the

Emergency Number to go through the PSTN
Gateway that references the Emergency
Service Provider
Note: It does not differ from any other Voice
Route
PS>New-CsVoiceRoute Name "EmergencyCallsRoute" -NumberPattern "^\
+911$" PstnUsage @{add="EmergencyCallsUsage"} PstnGatewayList
@{add="e911gw.fabrikam.net"}

Identity : EmergencyCallsRoute
Priority : 3
Description :
NumberPattern : ^\+911$
PstnUsages : {EmergencyCallsUsage}
PstnGatewayList : {PstnGateway:e911gw.fabrikam.net}
Name : EmergencyCallsRoute
SuppressCallerId :
AlternateCallerId :
 PIDF-LO Support on Trunk

To send location PS>Get-CsTrunkConfiguration

Service:PstnGateway:e911gw.fabrikam.net

information to the Identity :

E911 certified Service:PstnGateway:e911gw.fabrikam.net

OutboundTranslationRulesList : {}

provider, PIDFLO SipResponseCodeTranslationRulesList : {}

Description :

information must be ConcentratedTopology

EnableBypass
: True
: False

allowed to transit EnableMobileTrunkSupport : False

EnableReferSupport : True

through the Skype for EnableSessionTimer

EnableSignalBoost
: False
: False

Business Trunk MaxEarlyDialog

RemovePlusFromUri
: 20
: True
RTCPActiveCalls : True
Ensure that RTCPCallsOnHold : True
SRTPMode : Required
EnablePIDFLOSupp EnablePIDFLOSupport : True
ort on the Trunk is
set to True
Set-CsTrunkConfiguration -Identity "Service:e911gw.fabrikam.net"
-EnablePIDFLOSupport $true
Location Information Service

Part of Skype for Business Server 2015 web

services components:
Load-balanced within a cluster for high
availability
Precedence of matching client location
requests:
Basic service set identifier (BSSID) of Wi-Fi Access
Point
Switch/port from Link Layer Discovery Protocol
Media Endpoint (LLDP-MED)
Switch from LLDP-MED
Subnet

Media Access Control (MAC) match of

Location Information Service (continued)

Microsoft Windows PowerShell and GUI-

based administration
Follows NENA i2 reference architecture for
address validation
Follows IETF PIDF-LO standards with
extensions for location format
Other vendor/in-house LIS can be integrated
Configuring Location Information Server

Network identifiers are associated with street

addresses:
BSSID
Subnet
Switch
Switch/Port
The size of this data set will correspond to how
detailed the locations are and whether wireless is
within scope
You can use Windows PowerShell scripts to import
this data from .csv files
Address managementrelated administrative tasks
include:
Configuring the address validation service provider
Uploading validation credentials
Location Information ServerSubnet

Configure the LIS subnet

Note: There is no subnet mask. The client uses its
own IP/subnet mask to determine its subnet, and
then sends this in the Web Query to the LIS Web
Service
Set-CsLisSubnet -Subnet 172.16.20.0 -Description "Munchen"
-Location "Munchen"
-CompanyName "Fabrikam" -HouseNumber 2 -StreetName "Lindenstrasse"
-City "Munchen"
-Country DE

Publish the LIS configuration

Publish-CsLisConfiguration
Address Status

Every address added to the LIS database

should be validated by a Master Street
Address Guide (MSAG)
MSAG validation ensures that emergency
calls can be correctly routed
Client Location Request

Location Request includes the following

information:
Link Layer Discovery Protocol (LLDP) from Layer 2
connectionswitch and port IDs (if available)
Subnet
WAP BSSID (if available)
MAC address
LLDP is not supported for Skype for Business on
Windows 7 and earlier:
This makes it difficult to get detailed locations for wired
softphones
You will need to use an SNMP application to do this
LLDP is supported on Lync Phone Edition:
Switches may need upgrades to support LLDP
Automatically Acquiring a Location

Location database is globaleach LIS has all

defined locations
Client automatically initiates a location request
to its LIS:
Includes its network connectivity data
LIS location matches precedence:
Wi-Fi AP BSSID
LLDP-MED Switch/Port
LLDP-MED Switch
Subnet
MAC (If configured through Set-
CsWebServiceConfigurationMacResolverUrl)
If a location is not found, the request can be sent to
an external database
(Set-CsWebServiceConfiguration SecondaryLocationSourceUrl)
Using a MAC Address to Find a Location

Skype for Business does not natively map

the MAC address to Location information
LIS must resolve the MAC address into
switch/port for resolution:
Skype for Business client sends MAC address, and
then:
1) LIS queries external MAC Resolver through Web
Services
2) MAC Resolver returns switch/port
3) LIS uses switch/port to return location information to
client
This raises a question: What Is a MAC
Resolver?
It is an appliance provided by the Service
E9-1-1 Support for Remote Users

LIS Web Service is not exposed to external

users
How it works:
1) Users self-report their location
2) The call is automatically routed to the
Emergency Call Response Center (ECRC) with
location data
3) The ECRC confirms the location with caller,
and then transfers the call to the correct
Public Service Answering Point (PSAP)
Manual Location Entry

Location Policy field

LocationRequired
impacts user experience:
Location Required = No:
User not prompted
Location Required = Yes:
UI highlights location with X and !
for emphasis
Can be dismissed without warning
Location Required = Disclaimer:
UI highlights location with X and !
for emphasis
Disclaimer shown when dismissed
Only shows during sign in. No
effect on call
Emergency Dialing

User dials emergency number

Call goes straight through to the correct
PSAP:
If the location has been validated
Security Desk Integration

IM automatically established between

emergency caller and security desk:
Location information is contained in the
conversation window
An E.164 number can be bridged onto
emergency calls
One-way/two-way
Partner is responsible for initiating conference
Microsoft Official

Course
PSTN Integration
Background Definitions

Public switched telephone network (PSTN)

Private Branch eXchange (PBX)
Voice over Internet Protocol (VoIP)
Session Initiation Protocol (SIP)
Internet Telephony Service Provider (ITSP)
Uniform Resource Identifier (SIP URI)
Multiple Points of Presence (MPOP)
UCOIP

Skype for Business Certification Program

Testing and qualification of third party solutions for interoperability with
Microsoft UC
Independent testing by third party labs based on standards based open
documentation
SIP trunking providers supported with Lync Server 2013 will be supported
with Skype for Business

Supporte
Qualified Qualified
d
Gateway PBX
PBX
Typical Legacy Enterprise PBX

PSTN

Numbering Plan
31-20-500 1000 to
+31-20-500 1999

Class of Service
Class of Service
Outbound only
Inbound/Outbound
Local, National,
Local, National
and International

Dialing Habits
4 digit internal extensions
9 for an outside line
3 digits + extension for other locations
 Decision 1: Legacy PBX integration

Connect Skype for Business

Connect Skype for Busines
directly to the PSTN to the Legacy PBX

PST PST
N N
Decision 2: POTS/TDM or SIP Trunking

Connecting through a Gateway

PSTN

Connecting through SIP Trunk

PSTN

SIP TDM
 Direct Connection Through a Gateway

A gateway is a physical device that

connects two incompatible networks
The gateway translates signaling and
media between Skype for Business (SIP)
and the PSTN
Use supported gateways (UCOIP)
Skype for Business
Qualified
Skype for BusinessMediation PSTN
Pool Server
Gateway

PSTN

SIP TDM
 Direct Connection Through SIP Trunking

IP connection that establishes a SIP

communications link between your
organization and an Internet telephony
service provider (ITSP) beyond your firewall
Use supported SIP Trunking Provider
Session
(UCOIP)
Skype for Business
Border Qualified
Mediation
Skype for Business Controller IP-PSTN
Server
Pool (SBC) Gateway

PSTN

Enterprise Network VPN ITSP Network

SIP TDM
User Configuration

Skype for Business

and PBX phone
numbers can be
the same

Configuration is
roamed for MPOP
endpoints, saving
state of
CallViaWork at the
endpoint & whether
its in use.
Connecting Through PBX by Using SIP

PSTN
Skype for Business
Skype for Business Mediation
Server
Pool

Qualified or
supported
IP-PBX

SIP TDM IP endpoint

Connecting Through PBX by Using a
Gateway

Qualified PSTN
IP-PSTN
Gateway
Skype for Business
Pool Skype for Business
Mediation
Server

TDM or
unsupported
PBX

SIP TDM IP endpoint

PSTN Sizing

1. In replacement scenarios, existing call

volume is known
2. Account for new behaviors and features:
Simultaneous ringing
PSTN conferencing
Dial-in audio conferencing
Mobile users

.Use Erlang B calculations

when appropriate
Inter-Trunk Routing - Overview
Skype for Business Server 2015 supports call routing from
an incoming trunk to an outgoing trunk to provide routing
functionalities to other telephony systems
A possible alternative for PBX Integration scenarios

By enabling inter-trunk routing, the following routing paths

(among others) are enabled:
Incoming PSTN calls to an IP-PBX system via Lync
Outgoing IP-PBX calls to a PSTN network via Lync
Outgoing IP-PBX calls to another IP-PBX system via Lync
Inter-Trunk Routing Description
Skype for Business Server 2015 allows to the associate a set of PSTN usages
on an incoming trunk to determine a call route to an outgoing trunk

These PSTN usages are used to determine destination for incoming call on a
trunk, if the call cant be terminated locally
No local client or other entity is found (essentially, the RNL fails)
No match to CallPark range or Unassigned Numbers range

Inter-trunk routing call authorization scope is at the trunk level

The same call authorization applies to all incoming calls arriving via the trunk, that
cant be terminated locally on a client

Media bypass in inter-trunk routing calls is supported

 Inter-Trunk Routing

IP-PBX to IP-PBX
Skype for
Peer to Peer Routing
Business
without Skype for Business
Server 2015
Inter-Trunk
Routing
 Inter-Trunk Routing Signaling and Media
Flow
Routing of IP-PBX calls to PSTN via Skype for
Business
Incoming call from the PBX trunk
RNL fails
No match to Unassigned Numbers nor Call
Park ranges
Validate incoming trunk associated PSTN
usages
Determine a route
Apply outbound translation rules
Route to outgoing gateway trunk
Media-bypass possible if IP-PBX supports it
Routing of IP-PBX calls to another IP-PBX system
via Skype for Business
Incoming call from the PBX trunk
RNL fails
No match to Unassigned Numbers nor CallPark
ranges
Validate incoming trunk associated PSTN usages
Determine a route
Apply outbound translation rules
Route to outgoing PBX trunk via Lync or Skype for
Business
Media-bypass possible if both IP-PBX support it
Configuring Inter-Trunk Routing
Use the Skype for Business Management
Shell
New-CsVoiceRoute -Identity RedmondRoute -PstnUsages
Configure a voice route
@{add=Redmond"}
-PstnGatewayList @{add="PstnGateway:redmondgw1.contoso.com"}

Add a PSTN usage to a trunk configuration:

New -PSTNUsages property has been added
Set-CsTrunkConfiguration Identity TrunkId -PstnUsages
to CSTrunkConfiguration
@{add=Redmond}

Or use the Skype for Business

Control Panel
Mediation Server

Collocation vs. Standalone

Collocation can offer significant server count
reduction
Standalone may be preferable for network zone
placement or workload isolation
Media Bypass and Scalability
Scale based on hardware and transcoding mix
For planning, do not count calls with media
bypass
Pool vs. Single Server
Can gateway or SIP trunk support DNS load
balancing?
Media Bypass
Location Based Routing

Hyderabad
Skype for Business Pool

Bangalore Hyderabad
Gateway Gateway

Skype for Business Skype for Business

Mediation Mediation
Server Server

PSTN
Interworking Routing-History

Lync Server 2010:

Multiple PSTN gateways can be associated with the
same Mediation Server pool (1:N); a single PSTN
gateway is associated with a single Mediation Server
pool; a single SIP listening port on the Mediation Server
and on the gateway is used in the association.
Lync Server 2013:
Introduces M:N Interworking routing. A particular PSTN
gateway can be associated with multiple Mediation
Server pools or the same Mediation Server pool with
multiple unique associations.

Skype for Business Server 2015:

Introduces M:N Interworking routing. A particular PSTN
gateway can be associated with multiple Mediation
Server pools or the same Mediation Server pool with
multiple unique associations.
Trunk and IP-PBX Interworking

Multiple trunks between a Mediation

Server and PSTN gateway can be Mediation IP-PBX

defined to represent IP-PBX SIP

Server

Port A Trunk 1 Port A1

termination Port B Trunk 2 Port B1

Each trunk will be associated with

the appropriate route for outbound Port n Trunk n Port n1

calls from Mediation Server to IP-PBX

For inbound calls, per-trunk policy
will be applied
Trunk configuration will be scoped
globally or per trunk; similarly, dial
plan can be scoped per trunk
Representative Media IP is a per-
trunk parameter, allowing for Media
Bypass
 Trunk and IP-PBX Interworking-Real Life

Trunk 1: MS10 to
PBX01
PBX01 port: 5060
Mediation Server Signaling IP: PBX-1
(MS10) Media IP: MTP-1

IP-PBX/Gateway
(PBX01)

Trunk 2: MS10 to
PBX01
PBX01 port: 5061
Signaling IP: PBX-1
Media IP: MTP-2
Configuration Details

Topology Builder:
Define the PSTN Gateway and Trunks
Define the MTP as the Alternate Media IP address
Use different gateway listening ports for each trunk
Publish the topology
Windows PowerShell:
Identify the trunk IDs
Use Windows PowerShell to configure media IP
addresses for the remaining trunks
Verify the media IP address for the trunks
Trunks and Resiliency

Mediation
Server MS1
Port A Gateway GW1

Port B
Trunk1

Trunk2
Mediation
Server MS2
Gateway GW2
Port C

Port E
Trunk3
Multiple Sites to the Same Service Provider

Lync Server 2010:

Virtual gateways must be defined to allow
connectivity from multiple Mediation Server pools
to the same Session Border Controller (SBC) FQDN
SBC
Virtual gateway FQDNs all resolve to the same IP PSTN
sbc1.provider.com

address
TLS cannot be used because the SBC certificate
Trunk 1 Trunk 2

does not contain the virtual gateways name MPLS

Gateway-specific inbound policies cannot be Site 01 Site 02

Mediation Pool Mediation Pool
applied when virtual gateways are used (RNL of
the IP-address does not resolve to virtual gateway)
Lync Pool

Lync Server 2013 & Skype for Business:

Separates PSTN gateways and trunks
Enable you to connect multiple trunks to one
gateway
Enables the use of TLS
Allows for gateway-specific inbound policies
M:N Interworking Interworking-Trunk Definition
Auxiliary Calling Information
Skype Call
Incoming for Business Server 2015
to
+1 (989) 555
PSTN Phone
0200
+1 (999) 555
2001 User Bob
+1 (989) 555
Simultaneous
0200
Ring:
+1 (999) 555
1000
PSTN Phone INVITE
INVITE sip:+19995551000@192.168.1.41;user=phone
sip:+19995551000@192.168.1.41;user=phone SIP/2.0
SIP/2.0
+1 (999) 555 FROM:
FROM: sip:+19995552001@contoso.com;user=phone
sip:+19995552001@contoso.com;user=phone
1000 TO:
TO: sip:+19995551000@192.168.1.41;user=phone
sip:+19995551000@192.168.1.41;user=phone
HISTORY-INFO:
HISTORY-INFO:
sip:+19895550200@se01.contoso.local;user=phone
sip:+19895550200@se01.contoso.local;user=phone
ms-retarget-reason=forwarding,
ms-retarget-reason=forwarding,
sip:+19995551000@se01.contoso.local;user=phone
sip:+19995551000@se01.contoso.local;user=phone
P-ASSERTED-IDENTITY:
P-ASSERTED-IDENTITY:
<tel:+19995552001>
<tel:+19995552001>
SIP Header sent to 19995551000
Fast Failover and Options Polling

Gateway Log
1d:0h:12m:15s OPTIONS sip:192.168.1.41 SIP/2.0
FROM: <sip:se01.tailspin.local:5068;transport=Tcp;ms-opaque=6b773cd98097b3f8>;
epid=BE80B79150;tag=cdee90d70
TO: <sip:192.168.1.41>
CSEQ: 3 OPTIONS
CALL-ID: 598db21985cb4d38a5e89a410987464a
MAX-FORWARDS: 70
VIA: SIP/2.0/TCP 192.168.1.52:59546;branch=z9hG4bK3b462b11
CONTACT: <sip:se01.tailspin.local:5068;transport=Tcp;maddr=192.168.1.52>
CONTENT-LENGTH: 0
USER-AGENT: RTCC/5.0.0.0 MediationServer

Mediation Server Log

1d:0h:12m:15s SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.52:59546;branch=z9hG4bK3b462b11
From: <sip:se01.tailspin.local:5068;transport=Tcp;ms-
opaque=6b773cd98097b3f8>;epid=BE80B79150;tag=cdee90d70
To: <sip:192.168.1.41>;tag=1c1952373857
Call-ID: 598db21985cb4d38a5e89a410987464a
CSeq: 3 OPTIONS
Contact: <sip:192.168.1.41:5060;transport=tcp>
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-/v.5.80A.053.005
 Call-Routing Reliability-Lost Connection

Skype for Business Mediation Qualified

Route Policy: Server Pool Gateways
For the example session only,
Gateway (GW-01 and GW-03 options
in that order) can be used
GW-01
options

503 response MS-01

options GW-02

options

Front-End Server MS-02 GW-03

SIP Configured Trunk Control messages
 Call-Routing ReliabilityGateway Down

Skype for Business Mediation Qualified

Route Policy: Server Pool Gateways
For the example session only
Gateway GW-01 and GW-03 options
in that order can be used
GW-01
options

504 response MS-01

GW-02

options

Front-End Server MS-02 GW-03

SIP Configured Trunk Control messages
 Call-Routing Reliability and Retries

Skype for Business Server 2015

(FE) (MS) (GW1) (GW2)

Invite (trunk 1)
10-sec timer-1: starts
183 response
Timer-1: continues Failed Connection
Cancel (trunk 1)
Timer-1: expires
Invite (trunk 2)
10-sec timer-1: starts
183 response
Timer-1: continues
Invite

18x response 18x response

Timer-1: stops
Call-Routing ReliabilityNext-Hop Proxy

The Mediation Server tracks its next-hop

proxy and backup next-hop proxy by
sending out periodic options polls:
Backup next-hop proxy is defined by pool
pairing
If the primary next-hop proxy is found to
be down (failure to answer to five options
polls in a row), new invites from gateways
are sent to the backup next-hop proxy
Additionally, a 10-second timer is used for
incoming calls, so if the primary next-hop
proxy is used for a call and no SIP response
is received within this time, the call is
 Voice Routing Coexistence
Home Mediation
Outbound Calls Server Server
Supported

Skype for Business 2015 2015 2015 Yes

Skype for Business Server 2015 2013 Yes
2015 and
2013 2015 Yes
Lync Server 2013
Skype for Business Server 2015 2010 Yes
2015 and
2010 2015 No
Lync Server 2010
Next-
Mediatio Home
Inbound Calls n Server
hop
Server
Supported
Server
Skype for Business Server
2015 2015 2015 Yes
2015
Skype for Business Server 2015 2015 2013 Yes
2015 and
2013 2013 2015 Yes
Lync Server 2013
Skype for Business Server 2015 2015 2010 Yes
2015 and
Lesson 5: Call via Work - Expanding Voice interoperability to the PBX
phone

Skype Voice for PBX Users

End-users can make voice calls using any PSTN phone, including existing PBX end
Leverages existing Direct SIP connectivity between PBX systems and Skype for Bu

User Experience
Server dials out to PSTN or Deskphone number to connect user, then connects with far-end dest

Features
Presence update & call control from rich client
Mid-call control capabilities preserved on PBX phone
 Call via Work - Components

1. User instantiates call from Skype rich

client
2. Skype for Business Server places call
Destination 6 PSTN to users PBX station set (or to any
other PSTN phone number)
5 3. PBX routes call and local user answers.
Skype Server Pool
4
PBX 4. When Server sees this call answered,
2
places far-end call. Here the server will
Local call use PBX users DID as ANI
1 3
5. PBX routes call out to PSTN with users
Far-end call
DID (or to any other local PBX
endpoint)
Skype for Business PBX Station 6. Far-end call answers & call is
established
with client acting as control channel
Establishing a call
Mid call controls
Adding Modalities to a Skype for Business
call
Adding Modalities (IM)
 Multiple Calls

User warned on accepting/placing 2nd call

Lose control of the 1st call from client when second
call is started.

Remote participant activity

Remote participant may accept or place another call
from/to someone
This will make the call on PBX Phone go on hold for
the local user,
Conversation Window will not update to show the
accurate status of the call.
 Ending a call

Placing the receiver of the PBX phone on the

handset
Clicking the hang-up button
Close out (x) on the Conversation Window
 Conversation History
Works as expected
The initial inbound calls are not shown in
Conversation History view.

Inbound missed calls

PBX or Gateway should support Reason header Call
completed elsewhere in the CANCEL message
If PBX does not send this Reason header, Server will
treat incoming call as missed.
 Meetings
Client will prompt for meeting join preference

Dialog auto-populated

Focus dials out to users

CvW configured number .
Click to Join
Meet Now & Ad-hoc Group Call
Ad-hoc incoming group calls
 Inbound Calls

Call via Work is Outbound

Only

Inbound experience to
both client & phone
achieved when Skype is
first in line & forwarded
with Call FW settings

When PBX is first in line,

inbound call will land only
on desktop phone.
Presence

Scenario Behavior

Outbound CvW Call Presence will change to In a Call

Presence will change to In a Conference

Outbound Meet Now / Group Call
Call

Inbound CvW Call Answered on PBX No change to presence

Inbound CvW Call Answered on Skype Presence will change to In a Call

Presence will change to In a Conference

Inbound Meet Now / Group Call
Call
Policy and User configuration
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