Cisco Cucm To Webex Calling
Cisco Cucm To Webex Calling
Cisco Cucm To Webex Calling
Contents
CONTENTS .............................................................................................................................. 2
INTRODUCTION ....................................................................................................................... 6
TRANSITION .......................................................................................................................... 12
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Contents
REFERENCES ......................................................................................................................... 64
LICENSING ..................................................................................................................................64
NETWORK REQUIREMENTS.............................................................................................................64
PHONES .....................................................................................................................................64
USERS ........................................................................................................................................65
CALLING CONFIGURATION .............................................................................................................65
LOCAL GATEWAY.........................................................................................................................65
DESIGN OVERVIEW .......................................................................................................................66
DCLOUD LAB...............................................................................................................................66
COLLABORATION TRANSITIONS ......................................................................................................66
GLOSSARY ............................................................................................................................ 67
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What’s New in This Guide
© 2021 Cisco – CTG TME Collaboration Transitions – Calling: Transitioning from Unified CM to Webex Calling PAGE 4
What’s New in This Guide
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Introduction
Introduction
Target Audience
This transition deployment guide is intended to be used by teams or individuals with
experience configuring and administering Cisco Unified Communications Manager
(Unified CM) and Cisco Endpoints including IP desk phones, video devices, and
Jabber soft clients. There are links to product and support documentation throughout
this document to assist.
Overview
With the growth of cloud-delivered collaboration services, more and more customers
are looking to move their existing collaboration workloads to the cloud given the
promises of reduced total cost of ownership, simplified management, continuous
feature delivery, increased scale, and superior reliability inherent in cloud-based
services. As customers look to make the transition from on-premises to cloud
collaboration services, it’s important for them to understand what the transition entails
and the steps required to make the transition.
The purpose of this document is to provide deployment guidance for customers
specifically looking to transition from on-premises Cisco Unified CM calling to Webex
Calling in the cloud. This deployment guide assumes that the reader has a basic
understanding of the calling transition between Unified CM and Webex Calling
including what changes when making this transition and what the differences are when
moving the calling workload from on-premises to the cloud. Before proceeding ensure
you have reviewed and are familiar with the information available in the transition map
Calling: Transition from Unified CM to Webex Calling available at
https://www.cisco.com/c/dam/en/us/td/docs/solutions/PA/mcp/TDM_CALLING_Unifi
ed_CM_to_Webex_Calling.pdf. This transition map document provides information
about the changes and differences of this transition.
As shown in Figure 1, a typical deployment includes different collaboration
infrastructure components on the network, a call control platform, and an edge
platform, hardware and software endpoints, and in some cases even conferencing and
scheduling platforms. In the Cisco architecture this would include Cisco Unified CM for
call control, Cisco Expressway for remote access and business-to-business (B2B)
edge services, Cisco Meeting Server / Cisco Meeting Management for on-premises
conferencing, Cisco Unity Connection for voice messaging, and user-facing hardware
(Cisco IP Phones, Webex DX and Room) and software (Cisco Jabber) IP-based
endpoints. These components may vary slightly in some environments, but this is the
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Introduction
starting point for the transition described in the rest of this document.
Note: The architecture shown in Figure 1 is based on the Preferred Architecture (PA)
for Cisco Collaboration Enterprise On-Premises Deployments. For more information on
the Enterprise On-Premises PA, refer to https://www.cisco.com/go/pa.
Table 2 lists the key elements of the on-premises architecture prior to transitioning to
Webex Calling in the cloud:
Product Description
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Introduction
As illustrated in Figure 2, customers who have an on-premises call control with Unified
CM and desk and video IP endpoints have a choice of transitioning the architecture
toward a Webex Calling cloud architecture.
The decision needs to be made based on customer’s functionality requirements.
Customers that have the following requirements should consider carefully before
making this decision and may ultimately decide to keep call control on-premises:
Phone models other than Cisco 7800 and 8800 IP phone series.
Complex or numerous integrations with other on-premises systems / solutions.
Complex dial plan and/or highly granular classes of service.
Calling within deployment is predominately on-net within a single site.
Restrictive, limited, or unreliable Internet access.
Stringent data privacy and ownership policies.
Compliance requirement for on-premises or in-country media recording and
storage.
Requirement for more than 250 concurrent PSTN and cloud / on-premises
interop calls per Local Gateway (LGW).
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Introduction
Customers who wish to start leveraging Cisco cloud calling services should consider
Webex Calling. This cloud calling service allows the customer to leverage the Webex
global architecture for scale and connectivity. Participants on the corporate network
and remote participants outside the corporate network can communicate using IP-
based hardware endpoints or desktop or mobile soft client applications.
This document focuses on customers with Cisco Unified CM call control deployments
that want to understand the general steps, considerations, and requirements for
enabling Webex Calling deployment as depicted in the next section.
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Core Components
Core Components
Roles of the Components Involved
The target architecture for this migration includes several new components. This
includes the Webex Calling service for cloud-based calling, Webex App, Cisco
Directory Connector for identity integration, and Local Gateway IOS-XE router for
PSTN access as well as on-premises to cloud calling integration. Cloud Connected
PSTN (CCP) facilitated by a provider partner is another option for PSTN access.
As shown in Figure 3, the new components (Webex Calling, Directory Connector, and
Local Gateway) are added to the existing on-premises deployment.
Table 3 lists the new elements of the architecture after transitioning to Webex Calling.
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Core Components
Product Description
Cloud-based call service delivered from the Webex
Webex Calling platform and providing endpoint registration and call
routing
Windows application running on a Windows domain
Cisco Directory machine providing identity synchronization between the
Connector enterprise Active Directory and the identity store of the
Webex organization.
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Transition
This section covers the pre-transition preparation steps, the transition implementation
steps, and the post-transition steps to be considered for this workflow transition.
This document describes a phased transition in two parts. As shown in Figure 4, the
initial transition phase (Phase 1) results in a hybrid deployment with dual call control
where some devices are transitioned to cloud calling and other devices maintain on-
premises call control for registration and call routing. The final transition phase (Phase
2) results in a pure cloud calling environment where all devices have been fully
transitioned to cloud call control. How long an organization takes to transition to cloud
calling fully will vary based on the deployment in question. In some cases, organization
may make the initial transition and remain in the hybrid dual call control phase (Phase
1) for an extended period (months or even years) while in other cases an organization
may fully transition to cloud calling (Phase 2) in a very short period of time (days or
weeks). This document is intended to cover both partial (Phase 1) and full transitions
(Phase 2).
Note: It is possible that some organizations may maintain a hybrid dual call control
deployment indefinitely with no plans to ever fully transition to cloud calling.
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Licensing
Understanding the current licensing structure of an existing deployment is a key
consideration when preparing to migrate to Webex Calling. Perform a license
assessment of the following areas of your existing Cisco on-premises solution.
Platform
The ability to fully articulate what is currently licensed on your core platform
will be critical when working with your account team or partner to determine
the best path to Flex licensing. Webex Calling is licensed using Flex licensing.
For more information on Flex licensing refer to the Cisco Collaboration Flex
Plan information available at https://www.cisco.com/c/en/us/products/unified-
communications/collaboration-flex-plan/index.html.
Devices
Determine what license category your existing and planned new devices will
belong to with Webex Calling. Webex Calling licensing station types include
knowledge worker, basic user, and non-user assigned devices for common
areas. For more information on Webex Calling device licensing refer to the
data sheet available at
https://www.cisco.com/c/en/us/products/collateral/unified-
communications/webex-calling/datasheet-c78-742056.html.
Local Gateway
Because Cisco Unified Border Element (CUBE) is required for PSTN access for
this transition, CUBE licensing must also be considered. CUBE licensing
considerations are covered later in this document.
Deployment Sites
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The number and types of sites (large central, regional, branch, and so on) within
your existing deployment should be considered when planning this transition. A full
understanding of the existing deployment sites will aid in strategically planning for a
successful transition particularly when it comes to determining what sites to
migrate and in what order. Understanding in detail dial plan requirements
(numbering, dialing habits, classes of restriction, and so on), site network
connectivity and bandwidth (Internet, WAN, LAN), and PSTN access (local or
centralized, IP or TDM) for each site will be critical when making migration
decisions. For more information on common deployment models and key
considerations, please refer to the collaboration deployment models information
available in the Collaboration SRND at
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucm/srnd/collab12/colla
b12/models.html.
Finally, it is important to understand the impact the transition to Webex Calling will
have on other collaboration services. Based on the objective of this document, the
general assumption is that if existing collaboration services outside of the calling
workload are to be maintained, then transition to the phase 1 hybrid deployment
mentioned above is expected. Examples of collaboration services that may require
hybrid deployment include contact center, meetings, paging, call reporting, and so
on. For more information about the transition of additional collaboration workloads
and services refer to the Collaboration Transitions documentation available at
https://cisco.com/go/ct.
Network Connectivity
Consider existing provider data connections (MPLS, SD-WAN, and so on) and
generally plan for direct Internet access at each location within your deployment.
Because you will be consuming cloud-based services, reliable Internet connectivity
with sufficient bandwidth is a base requirement. You should reconsider making this
transition if your organization locations’ Internet connection(s) are not generally
reliable with low latency and adequate up and downstream throughput.
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Table 4 shows the call types available with a Webex Calling deployment along with
the codec and maximum bandwidth required for each call type. As shown in Table
4 the required audio call bandwidth for each call type can be calculated using the
following general formula:
Codec -
Call Types Bandwidth Calculations
Bandwidth
Webex App / MPP1 Phone OPUS - Number of concurrent calls * 70 kbps
-> Webex App 70 kbps = Required network throughput
Webex App / MPP1 Phone OPUS – Number of concurrent calls * 70 kbps
-> MPP1 Phone 70 kbps = Required network throughput
Webex App / MPP1 Phone G.711 – Number of concurrent calls * 80 kbps
-> PSTN via LGW 80 kbps = Required network throughput
Webex App / MPP1 Phone G.711 – Number of concurrent calls * 80 kbps
-> PSTN via CCP 80 kbps = Required network throughput
Webex App / MPP1 Phone - G.722 – Number of concurrent calls * 80 kbps
> Enterprise via LGW 80 kbps = Required network throughput
Webex App / MPP1 Phone - OPUS – Number of concurrent calls * 70 kbps
> Webex Calling Voicemail 70 kbps = Required network throughput
1
Multiplatform Phone (MPP)
By summing the concurrent required network throughput per call type, the total
potential bandwidth requirement for a specific site can be determined.
All call legs are always anchored on the Webex Calling access SBCs. To determine
the required internet bandwidth for any given Webex Calling location not only the
inter-location calls need to be considered, but also intra-location calls and calls to
and from a Local Gateway at that location. For example, an intra-site call between
two MPPs would need up to 2 x 70 kbps full duplex on the location’s internet
access.
By summing the concurrent required network throughput per call type, the total
potential bandwidth requirement for a specific site can be determined.
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Number of
Call Types Concurrent Total Bandwidth
Calls
Webex App / MPP1 Phone
15 2 * 15 * 70 kbps = 2,100 kbps
-> Webex App
Webex App / MPP1 Phone
15 2 * 15 * 70 kbps = 2,100 kbps
-> MPP1 Phone
Webex App / MPP1 Phone
50 2 * 50 * 80 kbps = 8,000 kbps
-> PSTN via LGW
Webex App / MPP1 Phone
0 0 * 80 Kbps
-> PSTN via CCP
Webex App / MPP1 Phone -
15 2 * 15 * 80 kbps = 2,400 kbps
> Enterprise via LGW
Webex App / MPP1 Phone -
5 5 * 70 kbps = 350 kbps
> Webex Calling Voicemail
Note: All bandwidth values in Table 4 and Table 5 refer to IP bandwidth. Link
bandwidth is higher depending on WAN encapsulations.
Note: The bandwidth in Table 4 and Table 5 is for audio calls. For video call
bandwidth, Webex App and the MPP 8845/65 phones support H.264 video with
maximum resolution of 720p at a maximum bandwidth of 1,500 kbps per call.
However, the amount of bandwidth consumed at any point during the will fluctuate
based on variable bit rate inherent in video communications.
Webex Calling requires reliable Internet connectivity and offers global reach from
all the customer locations thus, eliminating the need for endpoint survivability. If an
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SRST like option for endpoints is critical to the existing deployment, then Webex
Calling should not be considered as a migration option.
Call Recording
Call Recording integration is between Webex Calling and Dubber (partner offering)
data centers and all recorded media is securely kept in the cloud. If compliance
and regulation require media be kept on-premises or in your country of
deployment, understand this option is not available as part of the call recording
architecture.
Voicemail
Voicemail is an integral part of the Webex Calling offer and integration with a
premise-based voicemail solution such as Cisco Unity Connection or Cisco Unity
Connection Express is not available. Further, there is no ability to migrate existing
Unity Connection voicemail messages or greetings to the native voicemail service
available with Webex Calling. Likewise, there is no migration of Unity Connection
call handlers and auto-attendant functionality to Webex Calling, however, the basic
auto-attendant functionality available with Webex Calling may be configured as a
possible replacement.
Currently, Webex Calling allows no more than 250 concurrent sessions from a
single Local Gateway, which by default becomes the session count limit for Local
Gateway based PSTN calls and Inter-site calls between Unified CM and Webex
Calling endpoints. If a Local Gateway deployment requires more than 250
concurrent calls, please contact Cisco Technical Assistance Center (TAC) to
request increasing this limit.
Any calls exceeding this limit are rejected with a “403 Forbidden”. The “show call
active voice” command can be run on the Local Gateway at any instance to
determine the total number of active calls.
Contact your Cisco account team to explore other deployment options if you need
a higher number of concurrent sessions.
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Poor network conditions between Local Gateway and Webex Calling access SBC
can limit the performance of the signaling connection leading to an even lower
concurrent calls limit. One-way latency between the Local Gateway and the Webex
Calling data center should not exceed 100 ms and the jitter should be less than
10 ms.
For more details on network requirements for Webex Calling, refer to the Cisco
Webex Calling Customer Network Minimum Requirements Service Guide available
at:
https://callinghelp.cisco.com/wp-content/uploads/2019/05/WC-Customer-
Network-Minimum-Requirements-Guide-v2.2_062019.pdf.
Customers can also use cscan.webex.com for network assessment which gives
information on customer’s network quality, how many calls can be established,
latency, and so on. For more information on the cscan tool, refer to the Use CScan
to Test Webex Calling Network Quality article available at
https://help.webex.com/en-us/y27bej/Use-CScan-to-Test-Webex-Calling-
Network-Quality.
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Each Webex Calling customer is provisioned on one of the four Webex Calling
instances. All provisioning information of that customer is stored in that Webex
Calling instance and the SIP signaling of all endpoints and Local Gateways
provisioned for that customer is tied to the Webex Calling instance the customer is
provisioned on. Because the initial Webex Calling region selection cannot be
changed later it is important to consider all relevant factors as part of the decision
process leading to the Webex Calling region selection. To avoid excessive
signaling round-trip delay it is important to decide early in the transition process
which Webex Calling instance should be used. Cisco recommends selecting the
Webex Calling instance which provides the lowest signaling round-trip times for
the largest number of users within the deployment.
Another factor to consider in the Webex Calling region selection is the country
availability of PSTN services provided by Cloud Connected PSTN (CCP) providers
available within that region. While during the transition period PSTN access for
Webex Calling devices must be facilitated via a Local Gateway to enable
interworking with Unified CM registered devices, after successful completion of the
transition, PSTN access for Webex Calling may be switched to CCP. At that point
the country availability of the CCP providers available within the Webex Calling
region becomes an important factor.
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Webex Calling country availability refer to the Where is Cisco Webex Available
article https://help.webex.com/en-us/n6fwepj/Where-is-Cisco-Webex-
Available#id_98285.
Each user in Webex Calling is provisioned with an extension. The extension length
is a fixed global setting: all extensions in a Webex Calling deployment have the
same length. Extension dialing can be used between Webex Calling users both
within a location and between locations. Abbreviated inter-site extension dialing
(the latter case) only works if the dialed extension is unique.
The dial plan described in the Preferred Architecture for Cisco Collaboration 14
Enterprise On-Premises Deployments, CVD does not support abbreviated inter-site
extension dialing. Instead, the Preferred Architecture for Cisco Collaboration 14
Enterprise On-Premises Deployments, CVD recommends establishing an enterprise
specific numbering plan by prefixing the extensions with a unique steering digit
followed by a fixed length site code and to use this number format for abbreviated
inter-site dialing.
Table 6 shows an example of three locations where the extension ranges of two
locations, NYC and RTP, are identical. Establishing an enterprise numbering
scheme with inter-site steering digit “8”, followed by a three-digit site code, and
the four-digit extension creates a non-overlapping abbreviated inter-site dialing
habit.
Table 6. Enterprise Numbering Example
Extension Enterprise
Site Site Code
Range Range
To allow for a smooth transition the set of dialing habits for users before and after
transitioning to Webex Calling ideally should be the same. To prepare for the
transition for each location the DID ranges and extension ranges (or abbreviated
intra-site dialing habits) need to be documented. Based on this information then
the inter-site steering digit needs to be selected.
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Table 7 shows an example of three locations and fixed length extension ranges.
Because overlapping dialing habits need to be avoided, it is important to make sure
that for any extension range the first digit of the range does not match the steering
digit for abbreviated inter-site dialing. If for example “8” is selected as the steering
digit for inter-site dialing, then no extension range in any site can start with “8”.
Typically, the extensions at a given location match the last few digits of the DIDs
assigned to that location. To avoid conflicts the first digit of the extension can be
changed. If, for example, DIDs in the +1 408 555 8XXX range are used in a given
location, then instead of using 8XXX as extension range 7XXX can be used for the
extensions in that site.
Table 7. Fixed Length Webex Calling Extension Ranges
Any seven-digit dial string dialed on a Webex Calling device using the US Webex
Calling dial plan always gets transformed to a full 10-digit national number. This
behavior makes it impossible to use seven-digit enterprise numbering schemes
with Webex Calling. If the existing enterprise numbering schema and the
corresponding abbreviated inter-site dialing habit has seven digits, then during the
transition to Webex Calling the numbering schema must be changed to a longer or
shorter form. The easiest way to achieve this is to add an additional padding digit
to the numbering schema. The new longer inter-site dialing schema only needs to
be adopted by users already migrated to Webex Calling. Users still on Unified CM
can continue to dial seven digits. The enterprise dial plan on Unified CM in this
case needs to make sure that abbreviated seven-digit dialing from Unified CM to
Webex Calling gets transformed to either +E.164 or to the abbreviated dialing
format deployed on Webex Calling. This needs to be done before sending the call
to the Local Gateway.
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for locations on Webex Calling, the site codes can easily be changed to three digits
by prefixing an arbitrary padding digit (“8” in the example) to the two-digit site
codes used in Unified CM so that inter-site dialing from Webex Calling phones
uses steering digit “8” followed by the padding digit “8”, the old two-digit site
code, and the four-digit extension. Users on Webex Calling don’t need to
remember new site codes; they only need to remember to use “88” as prefix for
inter-site dialing instead of “8” on Unified CM.
Table 8. Transitioning Seven-Digit Dialing
Enterprise Enterprise
Site Extensions Site Code Site Code
Range Range
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Information Comment
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When it comes to PSTN access for Webex Calling, it is important to understand the
following considerations:
PSTN is required for off-net calling and between enterprises.
Cisco never supplies the PSTN.
Only one PSTN option per location is possible:
o Cloud Connected PSTN (CCP).
o Cisco Local Gateway for on-premises PSTN and Unified CM integration.
Emergency call routing and lawful intercept are the responsibility of the PSTN
provider.
A Local Gateway is required to create a connection between Webex Calling and
Unified CM as long as Unified CM and Webex Calling coexist. This connection is
not only required to route all calls between Unified CM registered devices and
Webex Calling registered devices; it also provides PSTN access for all Webex
Calling devices. PSTN access for each location in Webex Calling can either be
facilitated by a CCP provider or via a Local Gateway. It is not possible to setup
Webex Calling so that inter-enterprise calls originating from within a given Webex
Calling location use a Local Gateway while PSTN calls use a CCP provider. This
limitation implies that during the transition PSTN access for both, Unified CM
registered devices and Webex Calling devices, needs to be through PSTN trunks
controlled by Unified CM. Only at the end of the transition, when no Unified CM
registered devices remain, PSTN access for all users can be moved to a CCP
provider. These steps are shown in Figure 6.
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At the on-premises side, the Local Gateway relies on a SIP connection or trunk to
Unified CM and in the case of IP-based PSTN connections a SIP trunk is also used
to communicate with the IP PSTN provider network. To connect to Webex Calling
in the cloud, the Local Gateway communicates via secure SIP TLS to the Webex
Calling service. This connection is anchored at a Webex Calling cloud-hosted
Access Session Border Controller (SBC) which serves as the gateway to cloud
calling services. In all cases, the Local Gateway relies on RTP or sRTP (encrypted
RTP) for all media connections.
In the case of CCP, cloud-hosted Peering SBCs serve as the SIP interconnect to
CCP provider partner networks. The Peering SBCs are responsible for all
integration aspects with the partner network.
On the other hand, a Local Gateway allows the customer to re-use an existing UC
enabled Cisco ISR or CSR1000v router (assuming the existing ISRs or CSR1000v
router are supported for Local Gateway and that the scalability of the existing
platforms is sufficient to carry the additional load of the Local Gateway role). This is
also preferable if they have a pending service contract with their current PSTN
provider. It also allows for PSTN interconnect in locations not supported by the
Cloud connected PSTN provider.
A single Local Gateway can be deployed and utilized by multiple Webex Calling
locations. However, only a single local gateway can be assigned to a location within
Control Hub. Further, if during the transition multiple split sites (some users within
the same location still on Unified CM and some already on Webex Calling) with the
same extension range exist and extension dialing from Webex Calling to Unified
CM is a requirement, then on Unified CM for extension dialed calls from Webex
Calling to Unified CM a location specific dialing context needs to be established to
enable Unified CM to differentiate between extension dialed calls from different
locations. This can be achieved by configuring dedicated Local Gateways for each
location requiring split site extension dialing so that the dialing context on Unified
CM can be established via calling search spaces inbound on the now location
specific trunks from the different Local Gateways. These Local Gateways do not
require dedicated CUBEs. All Local Gateways can be configured as “logical”
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instances on the same CUBE (or CUBE High Availability pair) by overlaying the
Local Gateway specific dial-peer sets within the CUBE configuration.
For all IP-based environments, customers have the option to deploy CUBE high
availability as Local Gateway for call preservation. CUBE high availability Layer 2
box-to-box redundancy uses the Redundancy Group (RG) Infrastructure protocol
to form an active/standby pair of routers. The active/standby pair share the same
virtual IP address (VIP) across the respective interfaces and continually exchange
status messages. CUBE session information is check-pointed across the
active/standby pair of routers enabling the standby router to immediately take over
all CUBE call processing responsibilities if the active router should go out of
service, resulting in stateful preservation of signaling and media.
Note: Check pointing is limited to connected calls with media packets. Calls in
transit are not check pointed, for example, Trying or Ringing state.
Refer to Figure 8 below which depicts a typical CUBE high availability as Local
Gateway setup.
Figure 8. CUBE High Availability with Local Gateway
The CCP option entails a static SIP trunk from the Webex Calling data centers to
CCP provider data centers. Redundancy is achieved by full mesh connectivity
between two Webex Calling data centers and two CCP provider data centers and
failover is automatic and transparent to end users. Figure 9 shows this high
availability architecture.
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When a customer Location is added within Webex Control Hub, the customer
administrator may select a CCP provider from the list of integration options. The
country of the customer location determines which CCP providers appear in the
drop-down list. Only CCP providers that support that country are displayed.
Before beginning the transition it’s important to inventory your existing hardware
and software endpoints. Having a complete list of phone types, models, and
quantities will ensure you can adequately plan for transitioning endpoints and
mitigating the impact to your deployment for those devices that cannot be migrated
to cloud calling. The inventory should be used to determine the endpoints to
transition, the endpoints to replace prior to the transition, and the endpoints that
may remain managed and registered to on-premises call control.
Desk Phones
For audio and video VoIP desk phones including Cisco IP Phone 7800 and 8800
series, Cisco Unified IP Phone 7900 series, and other personal desk endpoints,
only the 7800 and 8800 series endpoints are supported with Webex Calling. Prior
to transition these phones must be migrated to Multiplatform Phone (MPP) in order
to be transitioned to Webex Calling.
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All other desk phones will need to be replaced with 7800 and 8800 series
endpoints or must remain registered to Unified CM if you plan to maintain them.
Video Endpoints
Personal and room video endpoints including the Webex series, Webex Room
series, Webex DX series, and other hardware video endpoints are not supported
with Webex Calling. These devices must remain registered to on-premises Unified
CM or transitioned to the Webex platform (registered as Webex Devices) if cloud
registration is desired.
Note: When video devices are moved from Unified CM registration to Webex, the
URI for these endpoints will change as they are now cloud registered.
The underlying assumption for video endpoints in this deployment is that they are
running CE firmware and are shared endpoints used for point-to-point video calling
and multi-point on-premises or cloud-based conferencing.
Note: Phone models 8845 and 8865 are personal video endpoints and are
supported with Webex Calling.
Soft Clients
Cisco Jabber clients on desktop and mobile platforms are not supported with
Webex Calling. Depending on the deployment mode(s) implemented for Jabber
(IM-only, phone-only, and/or full UC modes), you may decide to maintain Jabber
registration to Unified CM and other on-premises services (Unified CM IM&P, Unity
Connection, Cisco Meeting Server). On the other hand, users may be transitioned
to the Webex app, the preferred cloud-based software client. Moving users to the
Webex app may be done prior to transitioning to Webex Calling to give users time
to familiarize themselves with the new application. Alternatively, users may be
migrated from Jabber on-premises services in phases after the transition to Webex
Calling begins.
To ease the initial transition to Webex app, consider moving users to the Webex
app with Unified CM calling, so they can begin using cloud messaging and meeting
services while at the same time enjoying the existing on-premises Unified CM
calling experience.
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Determine which users within the existing set of on-premises calling users will be
transitioned to Webex Calling. If all users will be transitioned, but the number of
users is large, it is a good idea to move users in groups in order to ensure that IT
staff and support personnel are able to handle the transition and any issues that
may arise. You should also allow some time to provide initial information and
training sessions to prepare users for this transition. User transition grouping can
be done based on a variety of criteria including the location or site users are
assigned to, users’ departments, or even user types (knowledge workers,
executives, mobile workers, and so on).
As an example, if users in the deployment are divided across three main sites, New
York (NYC), San Francisco (SFC), and Research Triangle Park (RTP), a user
transition plan may look like the plan outlined in Table 10 below.
Table 10. User Transition Plan by Site
Pre-transition
User Site / Post-Transition
Information and Transition Period
Location Support
Training Sessions
NYC
Week of April 1 April 15 – April 27 Week of April 29
(1,525 users)
SFO
Week of May 6 May 20 – May 31 Week of June 3
(1,600 users)
RTP
Week of June 3 June 17 – June 28 Week of July 1
(1,275 users)
Follow these transition steps to move from on-premises calling (Unified CM) to cloud
calling (Webex Calling):
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For more details, check the "Local Gateway for Webex Calling Ordering Guide" at
https://www.cisco.com/c/en/us/products/collateral/unified-
communications/unified-border-element/guide-c07-742037.html.
The first step in transitioning to Webex Calling is ensuring that there is connectivity
over the Internet between the on-premises network and the Webex cloud. Most
organizations do not connect directly to the Internet, but instead connect through
one or more firewalls. For this reason, it is important to understand the traffic flows
required between the on-premises network and Webex for Webex Calling.
Network and security administrators must understand these flows in terms of
direction, protocols, IP addresses, and port numbers so that corporate firewalls and
other network components can be configured to accommodate this traffic.
For information on the required flows including IP address, ports, and protocols
refer to the Port Reference Information for Cisco Webex Calling article available at
https://help.webex.com/en-us/b2exve/Port-Reference-Information-for-Cisco-
Webex-Calling.
Use this information to properly configure the firewall, proxies, and other network
infrastructure in the existing deployment to enable Webex Calling network flows.
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Before enabling directory integration between the corporate directory and the
Webex cloud identity store, the following set of summarized steps should be taken:
i. Add and verify organization domain(s).
ii. Convert existing users.
iii. Claim organization domain(s).
iv. Set up SSO.
v. Suppress automated user email invitation.
vi. Determine license assignment method.
To add a domain to your Webex organization use the Add Domain option under
Settings > Domain in Webex Control Hub (https://admin.webex.com/). Start by
entering the administrator domain and click Add. Then find the verification token
by selecting Retrieve verification token (available by clicking the ellipsis (…) next
to the domain name). This verification token must then be added as a DNS TXT
record to your DNS host. Once this is done click Verify next to the domain. If
successfully verified, you will see “Verified” next to your domain. Repeat this
process for each domain owned by your organization.
Note: You must add and verify the administrator domain first. Failure to do so
will result in administrator lockout.
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To claim a domain for your Webex organization under Settings > Domain in
Control Hub (https://admin.webex.com/), click the ellipsis (…) next to the
domain you added and verified previously and select Claim verified domain.
After claiming the domain, you will see “Claimed” next to your domain. Repeat
this process for each domain owned by your organization.
While optional, the use of single sign-on (SSO) is recommended to provide the
best end-user experience. The benefit of SSO is that a user can use a single
common set of credentials for authenticating to any Webex services as well as
other collaboration applications. With SSO, a user only must provide credentials
a single time per session in order to be authorized for any services they are
subscribed to.
To enable SSO for the Webex organization, from Control Hub navigate to
Settings and scroll down to Authentication, click Modify and then, select the
Integrate a 3rd-party identity provider. (Advanced). Next, click the Download
Metadata File button to download the file for import to your Identity Provider
(IdP).
At this point you will need to configure your IdP as appropriate and import the
metadata file you downloaded from Control Hub. Then, download or export the
metadata file from your IdP.
Return to Control Hub and on the Import IdP Metadata screen, drag and drop
the IdP metadata file or navigate to the file using the file browser. Next, under
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Finally, test the SSO setup by clicking the Test SSO Connection button. When
prompted enter valid SSO credentials to confirm SSO is working properly.
Assuming the test is successful, select This test was successful. Enable Single
Sign-On and click Next to complete the SSO configuration.
For more information on tested IdPs and enabling SSO for your Webex org refer
to the Single Sign-On Integration in Control Hub article available at
https://help.webex.com/en-us/lfu88u/Single-Sign-On-Integration-in-Control-
Hub.
Note: This setting toggle may only be turned on when SSO is enabled.
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4. Directory integration
The first step for transitioning users to Webex Calling is importing users from the
on-premises directory to the cloud identity store for the Webex Calling
organization. In addition to initially importing users to the cloud organization, regular
synchronization of user information from the corporate directory to the cloud
common identity store is imperative to ensure identity information is consistent in
both places.
The preferred method for importing and synchronizing users between the on-
premises corporate directory and the identity store in Webex is to use hybrid
directory service with Cisco Directory Connector. Directory Connector running on a
Windows domain server retrieves user information from the corporate Active
Directory and synchronizes this information to the cloud identity store using REST-
based APIs. The administrator can determine which users are synchronized and
what attributes are mapped between the on-premises and cloud directories based
on sync agreement configuration. This synchronization is performed at regular
intervals to ensure the cloud common identity store is up to date with any changes
to the corporate Active Directory environment.
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Next, click the Download and Install link to save the latest version of the connector
installation .zip file to the target Windows domain server. Unzip the file and run the
.msi setup file in the setup folder.
Note: For Windows server and other deployment requirements refer to the
Deployment Guide for Directory Connector available at
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cloudCollaboration/spark
/hybridservices/directoryconnector/cmgt_b_directory-connector-guide-
admins.html
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uid. The mail attribute plays a key role in Webex because it uniquely identifies
the user.
On the redundant Cisco Directory Connector, configure the same settings outlined
above, but use a unique name for the Connector Name setting (for example,
DIRSYNC2).
Once the base configuration is complete, click the Dashboard tab and then click
Sync Dry Run and click OK to confirm. A dry run validates your configuration and
ensures the expected user accounts will be synchronized prior to performing a full
synchronization. Review the dry run report to check for user or other sync
anomalies (especially mismatched users). Click Done to close the dry run report.
If no issues are found, next, perform a full sync by navigating to Actions > Sync
Now > Full. Click Yes to confirm the request.
After the first full directory sync has completed, you should configure the
synchronization schedule to ensure any changes to Active Directory (additions,
modifications, and/or deletions) are reflected in the Webex organization.
From the Directory Connector dashboard click the Configuration tab and under
Schedule specify the following:
The Incremental Sync Interval in minutes (for example, 30 minutes) which
determines how often an incremental sync is performed. An incremental sync
picks up user account adds, changes, and deletions in Active Directory.
Tick Enable Full Sync Schedule and select the time and day(s) of week to
perform a periodic full synchronization (for example, 11:30 PM on F(riday)). A
full sync picks up user avatar, attribute mapping, base DN, and LDAP filter
addition/changes as well as adds, changes, and deletions of user accounts.
Specify the Failover Interval in minutes (for example, 60 minutes) before the
secondary Directory Connector becomes primary and takes over incremental
and full synchronization. This setting applies for high availability deployments
with more than one Directory Connector (recommended).
For more details on deploying and configuring Directory Connector refer to the
Deployment Guide for Cisco Directory Connector available at
https://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cloudCollaboration/spark
/hybridservices/directoryconnector/cmgt_b_directory-connector-guide-
admins.html.
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Alternatively, rather than using Directory Connector, users may be manually added
to the Webex organization common identity store individually or in bulk using CSV
files. This is a manual process adding administrative overhead but may make sense
for small deployments.
Prior to setting up the Local Gateway for Webex Calling, ensure that Webex Calling
has been licensed and enabled for the organization, and the Local Gateway has
been onboarded within the Control Hub. For more details, refer to the Webex
Calling deployment guide.
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If any of the above are not supported dialing habits prior to the transition, for
example no abbreviated inter-site dialing habit exists, then they don’t necessarily
need to be introduced during the transition.
Figure 10 shows the best practice dial plan approach as described in the
“Preferred Architecture for Cisco Collaboration 12.x Enterprise On-Premises
Deployments, CVD”. Key characteristics of this approach include:
Single partition for +E.164 directory numbers.
Core routing based on +E.164 route patterns.
Normalization of all dialing habits to +E.164 using translation patterns.
Use of translation pattern calling search space inheritance (option “Use
Originator's Calling Search Space” set on translation patterns).
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For example, PSTN dialing (9+1+10D) from a device in SJC provisioned with line
calling search space “SJCInternational” will first get matched by the
“9.1[2-9]XX[2-9]XXXXXX” translation pattern which normalizes the called party
number to +E.164. The secondary lookup then uses the same calling search space
“SJCInternational” again (calling search space inheritance) and the +E.164-digit
string will either get matched by a +E.164 directory number in the “DN” partition or
by one of the PSTN route patterns in the “USPSTNNational” or “SJCPSTNLocal”
partition. Abbreviated intra-site and inter-site dialing habits are implemented by the
translations in the “ESN” and “SJCtoE164” partition. While the “ESN” partition is a
global partition (accessible for phones in all locations) the “SJCTOE164” partition is
only accessible for users in location SJC. This is assuming overlapping extension
ranges.
The first step to enable calling from Unified CM to Webex Calling is to make sure
that +E.164 destinations get routed accordingly. This can be achieved by adding a
“WebexCalling” partition to the dial plan, adding +E.164 route patterns for all
Webex Calling destinations to that partition, and finally adding the “WebexCalling”
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partition to all calling search spaces representing classes of service which need to
be able to reach Webex Calling. Creating a dedicated “WebexCalling” partition is
required to enable creation of a differentiated class of service for calls originating
from Webex Calling. To avoid call loops the inbound calling search space on the
trunk from the Local Gateway should not have access to the “WebexCalling”
partition.
To enable inter-site abbreviated dialing to the Webex Calling site the required
dialing normalization translation pattern “8121.2XXX” is added to the already
existing ESN partition. This is the same dialing normalization translation pattern
which also needs to be provisioned for a site with this site code on Unified CM: for
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The +E.164 route pattern matching on a the Webex Calling location’s DID range
can be provisioned while all DIDs are still hosted on Unified CM. The best match
pattern matching algorithm of Unified CM makes sure that when a number hosted
on Unified CM is dialed then the +E.164 directory number provisioned on Unified
CM is a better match than the wildcarded +E.164 route pattern pointing to Webex
Calling so that the calls get extended to a line on Unified CM and not sent to
Webex Calling.
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Equivalent to the +E.164 route pattern again the best match routing logic of Unified
CM makes sure that directory numbers using the enterprise number format are a
better match than the enterprise number route pattern in the “WebexCalling”
partition so that only enterprise numbers which don’t exist on Unified CM get
matched and sent over to Webex Calling.
Preparation
Prior to transitioning the first locations and users to Webex Calling the Local
Gateway configuration and Unified CM configuration needs to be completed as
described in these documents:
Configure Local Gateway on IOS-XE for Webex Calling:
https://help.webex.com/article/jr1i3r
https://help.webex.com/article/nqqzbk7
Also, in the Webex Calling administration make sure to configure the following
internal dialing settings:
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Figure 13 shows an example with steering digit “8”, three-digit site codes (the
steering digit is counted as part of the routing prefix length), and four-digit
extensions. The total length of enterprise number in this case is 8.
Transitioning a Location
To prepare for the transition of users from Unified CM to a Webex Calling location
these configuration steps need to be completed:
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Create location in Webex Calling with the setting collected based on the
information in Table 6 (see step 5. Inventory existing locations/sites in the
Pre-Transition Steps section).
Define site code and PSTN access code for location.
Add and activate phone numbers in Control Hub. Adding phone number in
Control Hub does not change Webex Calling call routing. Phone numbers
become active as soon as the number is assigned to user and a device gets
provisioned. All DIDs of users in the location to transition need to be added.
Provision the site-specific route patterns in the “WebexCalling” partition:
+E.164 and (if needed) enterprise numbers.
Because all users already exist in Control Hub (either through LDAP directory
integration with Directory Connector or manual individual or bulk provisioning), the
next step is to use bulk update (recommended) to enable appropriate users for
Webex Calling, assign them to a location, provision their phones and assign phone
numbers and extensions. The CSV template for this update can be downloaded
from Control Hub by clicking Manage Users and selecting the CSV Add or Modify
Users option. To avoid errors, you can also export all users, filter out the users to
modify and then update the settings only for a selected set of users.
Table 12 provides an overview of the CSV file columns relevant for the user
migration and the required settings. Quotes are used to indicate literal values;
insert the values without the quotes.
Table 12. User migration CSV settings
Column Setting
Extension extension
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To enable Webex Calling supplementary user and system features and services,
navigate to Control Hub to update users individually or in bulk (Bulk Edit Users) for
features like call forwarding, voicemail, and so on. Further, enable system features
like auto attendants, hunt groups, call queuing, and so on.
Once users provisioning is complete, the next step is to provision Webex Calling
devices and migrate existing phones as described in the next step.
In order to become Webex Calling cloud registered devices, the migration process
will use the cloud-based service to deliver the transitional and MPP firmware loads
to complete the migration. Since the transitional firmware loads are phone model
specific, the upgrade process will use common device selection criteria to make
sure the correct phones are migrated to the Webex Calling platform.
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The phone migration process is only supported on the 7800 and 8800 phones
(refer to Convert between Enterprise Firmware and Multiplatform Firmware for
Cisco IP Phone 7800 and 8800 Series Guide to verify supported phone
models and versions). Any phone device that does not meet the minimum
model version will need to be replaced with an MPP phone.
Additionally, any supported phone model must be on firmware version
12.5.1SR2 or later. If any phone is on a previous version, then the phone must
be updated to the latest Unified CM firmware prior to applying the transitional
firmware to the phone.
To check to make sure that no phone has a different load than the default
firmware load, in Unified CM, navigate to Device > Device Settings > Firmware
Load Information to load information on firmware loads for the devices
configured on the system as shown in Figure 14.
Selecting one of the hyperlinks on the Firmware Load Information page will
show you which phones may not be running the current default load. As
shown in Figure 15, any phone that has a name in the ‘Load Information’ field
indicates it is not using the default firmware.
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Clicking through on any phone in the list will allow direct access to any device
that needs to be upgraded to the model default firmware version.
The migration process should not continue until all phones to be migrated are
on version 12.5.1SR2 or later.
Note: This procedure was verified using 12.7.1 as the base firmware.
After ordering and downloading the licenses for migrating firmware, navigate
to https://upgrade.cisco.com/e2m_converter to upload your migration
licenses to the Cisco Cloud Upgrader service. Cisco Cloud Upgrader is a
service that allows customers to easily upgrade/migrate the software on Cisco
IP Phones so they can connect to Webex Calling.
Once the licenses have been uploaded, you will need to provision the phones
in Webex Calling.
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Table 13 provides an overview of the CSV file columns for device provisioning
and the required settings. Quotes are used to indicate literal values; insert the
values without the quotes.
Table 13. Bulk device operation settings
Column Setting
Type “USER”
Bulk provisioning should also be performed as described above for any new
phones added as part of this transition. This can be done now or later.
Note: Make sure to complete bulk provisioning for all phones to be migrated
before initiating firmware migration in the next step.
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Figure 16. Search by Device Type and Grouping Criteria of Device Pool
On the Update Phone page, as shown in Figure 17, select the Apply Config
setting. This instructs the phones to download the new configuration
information after executing the update.
In the Phone Load Name field, specify the correct model specific firmware for
the phone model that was originally selected. In this example, as shown in
Figure 18, the selected 8845/65 phone models will load the configured
transitional firmware load: sip8845_65.TLexE2m-11-2-3C-10.
Figure 18. ‘Phone Load Name’ Setting with Configured Transitional Firmware Load
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Next, the phone needs to be configured with the location to find the migration
firmware files. The Load Server value must be set to use the cloud
downloader. As shown in Figure 19, the cloud downloader URL is
cloudupgrader.webex.com.
Note: As shown in Figure 18 and Figure 19, the checkbox on the left side of
the Update Phones page is required for the Phone Load Name and the Load
Server entries. The Bulk Administration update job updates only the entries
that have a check box selected. In this example, only these two values are
getting updated on the selected phones.
At the bottom of the Updates Phones page, change the Job Description name
(in this example ‘Update Phones – 8865_E2M’) and select the Run
Immediately option in the Job Information section (see Figure 20). After
clicking Submit, the batch update will occur immediately, and all selected
phones will download the migration firmware from the cloud upgrade service.
Once the phones complete the loading of the interim load, they will contact
the cloud upgrader services, verify the device is entitled to migrate to Webex
Calling and download the latest MPP firmware. All of this happens
automatically once the interim load is successfully downloaded and the phone
reboots.
Each phone model has a different interim load, so the above process must be
repeated for each phone model type to be migrated.
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Table 14 shows the interim firmware loads for all supported models.
Table 14. Interim Firmware Loads for Eligible Webex Calling Devices
7832 sip7832.TLexE2M-11-2-3C-10
78xx sip78xx.TLexE2M-11-2-3C-10
8832 sip8832.TLexE2M-11-2-3C-10
8845/65 sip8845_65.TLexE2M-11-2-3C-10
88xx sip88xx.TLexE2M-11-2-3C-10
7832 sip7832.12-7-1-0001-393
78xx sip78xx.12-7-1-0001-393
8832 sip8832.12-7-1-0001-403
8845/65 sip8845_65.12-7-1-0001-393
88xx sip88xx.12-7-1-0001-393
After the phones download the latest enterprise firmware, repeat the process
above to convert the phones to cloud registered phones using the appropriate
.TLexE2M-11-2-3C-10 interim build for each phone model transitioned.
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Finally, to make sure that calls get routed from Unified CM to Webex, the
existing devices and directory numbers need to be deleted from Unified CM.
To delete the phones from Unified CM follow the steps described in the
Phones Deletions section of the Bulk Administration Guide for Cisco Unified
Communications Manager available at
https://www.cisco.com/c/en/us/support/unified-communications/unified-
communications-manager-callmanager/products-maintenance-guides-
list.html.
After deleting the devices from Unified CM, the now unassigned directory
numbers need to be deleted using the “Delete Unassigned Directory
Numbers” procedure in the Phones Deletions section of the Bulk
Administration Guide for Cisco Unified Communications Manager referenced
previously.
Routing from Unified CM to Webex Calling only becomes active after the
directory numbers have been deleted because the directory numbers always
are a better match than the wildcarded route patterns in the “WebexCalling”
partition.
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For US and Canada based telephony deployments that must provide enhanced
emergency calling solutions, Webex Calling may be augmented with Horizon
Mobility from RedSky (https://www.redskye911.com/horizon-mobility-for-webex-
calling). The Horizon Mobility solution meets all the regulatory requirements under
the recently enacted Kari’s Law and Ray Baum’s Act legislation (compliance
requirement anticipated in February 2021). The RedSky Horizon Mobility solution is
a subscription model and aligns with the subscription levels of Webex Calling end
users. RedSky’s Horizon Mobility is a cloud-based solution and has no reliance on
an on-premises application server managed by the customer (like Cisco
Emergency Responder).
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Figure 21. Cloud: Webex Calling Enhanced Emergency Calling with Horizon Mobility
The physical locations within the customers network are configured and managed
by the customer / partner in the Horizon Mobility portal. Any time a device sends a
location update via the HTTPS service, the device’s physical location will be
updated to be associated with the new upstream switch, switch port, wireless
access point, or IP Subnet.
When a user places an emergency call to 911 from a Webex Calling device (Figure
21, step 2), the call will be sent through a Webex Calling managed link to the
RedSky Horizon Mobility service (step 3). Redsky Horizon Mobility service will then
identify the location of the calling device and deliver the call to the appropriate
PSAP (step4). The PSAP that receives the call will also get the current physical
address of the calling device (step 5).
During the transition to Webex Calling the following is required to enable E911 with
RedSky Horizon Mobility:
Configure each physical dispatchable location in the Horizon Mobility Portal.
Assign each network element (switch, switchport, wireless access point, and
IP subnet or endpoint) to the physical dispatchable location.
Reboot all phones at the physical location to trigger the location registration
with Horizon Mobility using HELD.
Test emergency calling using the 933-test dial string to confirm that the
proper physical location of the calling device is processed.
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Call recording solutions provide a way to record audio and video calls that traverse
various components in a unified communications and collaboration solution. These
recordings can then be used by call centers and other enterprise operations for
various purposes such as compliance, transcription and speech analysis, or for
podcasting and blogging. Existing media forking call recording solutions for an on-
premises deployment, for example, CUBE controlled recording, Unified CM
network-based recording, and SPAN-based recording solutions are not supported
with Webex Calling.
The call recording solution for Webex Calling is provided by the partner Dubber
(https:///www.dubber.net). The Dubber solution enables recording of all calls
(Internal and PSTN) placed/received on the Webex Calling platform for replay and
management. Dubber integrates with the Webex calling platform in the cloud. Prior
to transitioning to Webex Calling, ensure your call recording compliance and
regulation is met by this architecture. For example, in certain countries, regulation
requires the recorded media to stay within the geographic boundary of that country
and that requirement may not be satisfied as Webex Calling platform and the
Dubber data center integration only exists in certain countries.
The following steps are required to deploy call recording in Webex Calling, which
should take place once the end users and devices have been migrated to the
cloud:
i. Partner must first establish an agreement and partner account with Dubber.
ii. Partner then sells the service to the Webex Calling customer.
iii. Partner enables call recording for the Webex Calling customer in Webex
Control Hub.
iv. The call recording license is assigned to an end user either by partner or
customer administrator.
v. The call recording feature for an end user is enabled within the Webex Calling
platform.
vi. Lastly, the partner creates the customer administrator and user in Dubber
portal using data from the Webex Control Hub.
Dubber call recording feature options for Webex Calling are configurable on a per
user basis by the customer administrator or partner and include the following:
Call recording designation.
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Once all endpoints and users are migrated to cloud calling the single purpose of
Unified CM is to act as transit between the PSTN gateways and Webex Calling via
Local Gateway. Removing PSTN gateways, Unified CM, and Local Gateway from
the picture by using Cloud Connected PSTN as PSTN access for all Webex Calling
users has several benefits including cost reduction and improved reliability. To
transition from Local Gateway based PSTN access to Cloud Connected PSTN
follow these steps:
Refer to the list of Cloud Connect PSTN providers and select from the available
provider(s) available for your organization’s location.
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Before switching PSTN access for Webex Calling locations to Cloud Connected
PSTN, connectivity to PSTN via the selected Cloud Connected PSTN provider
should be verified and validated. For this purpose, a test location needs to be
provisioned in Webex Calling with some test users provisioned in that test
location. PSTN access for this test location is then set to the Cloud Connected
PSTN provider before validating PSTN connectivity using the test phones. Upon
successful validation the test location can be deprovisioned.
To prepare for the cut-over to Cloud Connected PSTN a port order for all
numbers currently assigned to the PSTN trunk terminating on Unified CM needs
to be placed. All numbers need to be ported to the Cloud Connected PSTN
provider. To maintain inter-location reachability all numbers of all locations need
to be ported at the same time.
At date of the cut over PSTN access for all locations in Webex Calling needs to
be set to the Cloud Connected PSTN provider and inbound and outbound
connectivity should be verified.
Once all users have been transitioned to Webex Calling and all endpoints have
been transitioned to cloud registration (or have been decommissioned), update
appropriate on-premises infrastructure now that cloud calling is in use. Updates to
the infrastructure include:
Remove on-premises call control and messaging DNS SRV records from the
on-premises DNS server(s) including cisco_uds._tcp.<domain>,
cup_login._tcp.<domain>. These SRV records are no longer required for client
service discovery.
Remove edge-related DNS SRV records from the public DNS system including
collab_edge._tls.<domain>. These SRV records are no longer required for
client service discovery of collaboration edge services.
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Update all relevant DHCP scopes to remove option 66 and option 150
TFTP/boot server addresses. These scopes are no longer required for
endpoint call control configuration discovery and download.
Update/remove appropriate dial-peers in Local Gateway/CUBE that route calls
to and from Unified CM. These dial-peers are no longer required for on-
premises call routing.
Delete or remove all Unified CM and Expressway cluster node virtual machines
and/or servers. Repurpose compute resources and hardware as needed.
These resources are no longer needed for call control and edge services.
Delete or remove all Unity Connection cluster node virtual machines and/or
servers. Repurpose compute resources and hardware as needed. These
resources are no longer needed for voicemail and unified messaging services.
Clean-up: After migrating PSTN access to Cloud Connected PSTN Unified
CM, PSTN trunks, PSTN gateways, and Local Gateway can be
decommissioned.
For any existing on-premises e911 solution, delete any locations or numbers
that have migrated to Webex Calling and once full transition is complete,
remove application virtual machines or servers. Repurpose compute resources
and hardware as needed. These resources are no longer needed for
emergency calling and location services.
Update the physical dispatchable location and network element in Horizon
Mobility whenever changes occur. Common activities that require updates are:
o Network switch replacement.
o Wireless access point replacement.
o DHCP scope changes.
o Physical changes inside the building (if resolving to cubical/office).
o Physical office space expansion or contraction inside a building.
Admin and users can always follow the Webex calling service’s status on
https://status.broadsoft.com/.
During transition, admin can utilize existing tools to diagnose and resolve issues.
These include:
Unified CM traces.
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Transition
IOS-XE debugs.
Sniffer captures of network traffic.
Syslog.
Administrators can also open a service request with Cisco Technical Assistance
Center (TAC).
Webex Calling Analytics in Control Hub brings a new level of insight into your
Webex Calling deployments. With the enablement of this feature, historical data of
call usage and engagement is available in Control Hub, including media quality
records. The Webex Calling analytics are presented in Control Hub under
Analytics > Calling.
Engagement Analytics
Under Analytics > Calling > Engagement are historical engagement analytics for
your Webex Calling service (see Figure 22).
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As shown in Figure 23, the Calls and Call Minutes graphs contain details of all the
point-to-point calls made in the organization. The graphs are split into the following
categories, depending on the client used for the call:
Desktop –Webex App for Windows and Mac.
Browser –Webex App for Web.
Mobile –Webex App for iPhone, iPad and Android.
Device –Webex Room Device or Webex Board.
WXC Desktop –Webex Calling App for Windows and Mac.
WXC Mobile –Webex Calling App for iPhone, iPad and Android.
MPP –Webex Calling Multiple Platform Phone.
The Call Details table contains one entry for each call containing the following
information:
Name – Username.
Email – User email.
Start Date – When the call was made in GMT.
Duration – Duration of the call-in seconds.
Endpoint – Type of device or client used. WXC means Webex Calling app, and
MPP means Webex Calling Multiple Platform Phone.
Uaversion – User agent version if reported by the endpoint.
Call ID – A unique identifier for the call.
User ID – A unique identifier for the user.
As an example, if Alice calls Bob and both devices are within the same
organization, two records will be shown in this view, one for Alice and one for Bob.
Quality Analytics
The Quality Analytics tab allows you to view records for each call and use sliders to
filter calls based on quality statistics (see Figure 23).
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All statistics are collected from the devices/endpoints directly, and so the statistics
(packet loss, jitter, latency) reflect the experience of the call from the perspective
of the user’s endpoint.
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References
References
Licensing
Cisco Collaboration Flex Plan
https://www.cisco.com/c/en/us/products/unified-
communications/collaboration-flex-plan/index.html
Network Requirements
Prepare Your Environment for Webex Calling
https://help.webex.com/en-us/n4cprps/Prepare-Your-Environment-for-
Webex-Calling
Phones
Supported Devices for Webex Calling
https://help.webex.com/en-us/qkwt4j/Supported-Devices-for-Webex-Calling
Configure and Manage Webex Calling Devices, Add and Assign Devices in Bulk
https://help.webex.com/en-us/n9r1aac/Configure-and-Manage-Webex-
Calling-Devices#id_118912
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References
Users
Getting Started with Cisco Webex Control Hub
https://help.webex.com/en-us/nkhozs6/Get-Started-with-Cisco-Webex-
Control-Hub
Calling Configuration
Configure Webex Calling for Your Organization
https://help.webex.com/en-us/njvdjf2/Configure-Cisco-Webex-Calling-for-
Your-Organization
Local Gateway
Configure Local Gateway (IOS-XE) for Webex Calling
https://help.webex.com/en-us/b2exve/Port-Reference-Information-for-
Cisco-Webex-Calling
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References
Design Overview
Cisco Preferred Architecture for Cisco Webex Calling
https://www.cisco.com/c/dam/en/us/td/docs/solutions/CVD/Collaboration/hy
brid/AltDesigns/PA-WbxCall.pdf
dCloud Lab
Transitioning from Unified CM to Webex Calling Lab
https://dcloud-cms.cisco.com/demo_news/transitioning-from-unified-cm-to-
webex-calling-lab
Collaboration Transitions
Collaboration Transitions Program Page
https://www.cisco.com/go/ct
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Glossary
Glossary
The following tables provide a glossary of Unified CM concepts and constructs along
with a definition and information on potential alignment with Webex Calling.
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Glossary
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Users
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Phones
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Features / Functionality
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Glossary
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Glossary
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Glossary
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Glossary
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Operations
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Bulk Administration Tool Web Bulk Administration Tools are also available in
Webex Calling with some limitations:
Bulk User Import.
Bulk Device Import.
Bulk Device-User Assignment.
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Glossary
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