7. - Telephony & Infrastructure

Certificate Authorities

Navigation: Admin → Telephony → Certificate Authorities Last verified: Genesys Cloud Resource Center — March 2026

What Are Certificate Authorities?

Certificate Authorities (CAs) in Genesys Cloud are used to manage trusted digital certificates for secure TLS connections in telephony. Genesys supports two certificate types: Managed and Remote.

⚠️ This page applies primarily to BYOC Premises deployments. For BYOC Cloud TLS trunk configuration, refer to the BYOC Cloud TLS trunk transport documentation instead.

Certificate Types

Type Who Manages It Purpose Editable?
Managed Genesys Creates trusted TLS connections for the Edge and managed phones; allows remote SIP devices to trust secure connections to external trunks connected to the Edge No — cannot be added, edited, or deleted
Remote Customer (you) Imported CA that allows the Edge to trust a remote TLS endpoint such as an SBC or PBX Yes — can be added, edited, and deleted

ℹ️ There is only one managed certificate per organization. Genesys maintains it automatically.

Navigation

Task Path
Open Certificate Authorities Admin → Telephony → Certificate Authorities
Add remote certificate authority Certificate Authorities → Add
Edit remote certificate authority Certificate Authorities → select entry → Edit
Delete remote certificate authority Certificate Authorities → select entry → Delete

Required permission: Telephony > Plugin > All

Adding a Remote Certificate Authority

Step Action
Step 1 Navigate to Admin → Telephony → Certificate Authorities
Step 2 Click Add
Step 3 Choose import method: Upload from computer or Paste text from a file
Step 4 Upload the .crt file or paste the certificate text
Step 5 In Select Service for Use, choose the appropriate telephony service(s)
Step 6 Click Save Certificate Authority
Step 7 Test the secure TLS connection to the remote endpoint

UI Fields

Field Description
Type column Identifies whether the CA is Managed or Remote
Common Name Certificate authority common name
Add Certificate Authority Import method selector — Upload from computer or Paste text from a file
Browse Opens file browser to locate the .crt file
Enter Your Certificate Authority Text box for pasted certificate contents
Select Service for Use Associates the CA with one or more telephony services
Save Certificate Authority Saves the new or edited remote CA

Key Rules

Rule Detail
Managed CAs are read-only Cannot be added, edited, or deleted
Remote CAs are fully manageable Add, edit service associations, or delete as needed
Supported import formats .crt file upload or pasted certificate text
BYOC Premises scope This feature area is for BYOC Premises; BYOC Cloud has its own TLS trunk documentation

When to Use a Remote Certificate Authority

Situation Action
BYOC Premises Edge must trust a remote SBC or PBX TLS endpoint Import remote CA
Remote carrier presents a certificate signed by an internal/private CA Import remote CA
Managed phones require trusted TLS Use the Genesys-managed CA — no action needed
BYOC Cloud TLS trunk setup Do NOT use this page — use BYOC Cloud TLS trunk transport documentation

Troubleshooting

Issue Cause Resolution
Remote TLS endpoint not trusted Required remote CA not imported Import the correct CA and assign service usage
Cannot edit certificate authority Selected CA is of type Managed Managed CAs are read-only — only Remote CAs can be edited
Service still fails after import Wrong certificate or wrong service association Recheck the certificate chain and selected service(s)
Admin cannot access CA management Missing permission Grant Telephony > Plugin > All
Used wrong workflow for BYOC Cloud This page is for BYOC Premises Use the BYOC Cloud TLS trunk transport documentation instead

Quick Reference

Question Answer
What two certificate types exist? Managed and Remote
Who manages the Managed CA? Genesys
What is a Remote CA used for? Allows the Edge to trust a remote TLS endpoint
How can a remote CA be imported? Upload from computer or paste text from a file
Can Managed CAs be edited? No
Does this apply to BYOC Cloud? No — BYOC Cloud has its own TLS trunk documentation

See Also

Screenshots

Create New

DID & Toll-Free Numbers

Navigation: Admin → Telephony → DID Numbers Last verified: Genesys Cloud Resource Center — March 2026

What Are DID and Toll-Free Numbers?

DID (Direct Inward Dial) and toll-free numbers are the inbound phone numbers your organization uses. They must be added to Genesys Cloud as inventory before they can be assigned to a person, phone, or call flow.

Number Type Description
DID Geographic number with a local area code; used for direct user or department dialing
Toll-Free Non-geographic number (800, 833, 844, 855, 866, 877, 888); typically used for public-facing inbound access

Both DID and toll-free numbers are managed in the same workflow under Admin → Telephony → DID Numbers.

Two Main Areas

Tab Purpose
DID Ranges Add and manage blocks of DID or toll-free numbers
DID Assignments Assign individual numbers to a person, phone, or call flow; view and manage current assignments

Navigation

Task Path
Open DID Numbers Admin → Telephony → DID Numbers
Open DID Ranges DID Numbers → DID Ranges tab
Create a range DID Ranges → Create Range
Open DID Assignments DID Numbers → DID Assignments tab
Assign a number DID Assignments → select number → Assign
Unassign a number DID Assignments → select assigned number → Unassign

Step 1: Create a DID or Toll-Free Range

Numbers must be added as a range before they can be assigned.

Step Action
Step 1 Navigate to Admin → Telephony → DID Numbers
Step 2 Open the DID Ranges tab
Step 3 Click Create Range
Step 4 In DID Start, select the country and enter the first number
Step 5 In DID End, select the same country and enter the last number
Step 6 Enter the Service Provider (carrier/provider name)
Step 7 Save the range

Range Creation Fields

Field Description
DID Start First number in the range — country selector + number
DID End Last number in the range — same country as Start
Service Provider Carrier or provider name associated with this block

ℹ️ For a single number, enter the same value in both Start and End.

Step 2: Assign a Number

Once numbers are in inventory, assign them from the DID Assignments tab.

Step Action
Step 1 Open the DID Assignments tab
Step 2 Locate the desired number (search or filter by assignment status)
Step 3 Select the number
Step 4 Choose the assignment target type
Step 5 Select the specific Person, Phone, or Call Flow
Step 6 Save the assignment
Step 7 Test inbound routing

Assignment Target Types

Target Use Case
Person Assign a direct number to a specific user
Phone Assign a number to a specific device
Call Flow Assign a number to an inbound Architect flow (IVR / queue entry point)

Common Assignment Scenarios

Scenario Target
Employee direct inward dial Person
Main inbound IVR number Call Flow
Shared lobby or reception device Phone
Public-facing toll-free number Call Flow
Branded toll-free for a department Call Flow

Unassigning a Number

Select the assigned number in DID Assignments and choose Unassign. The number returns to available inventory and can be reassigned.

Troubleshooting

Issue Cause Resolution
Number not visible Range not created or not imported Recheck DID Ranges and provider data
Number cannot be assigned Already assigned or not in available inventory Filter by assignment status; unassign first if needed
Calls do not reach destination Wrong assignment target or downstream routing issue Verify the assignment target and its call flow/phone/user setup
Wrong user or flow receives calls Incorrect assignment Unassign and reassign correctly
Toll-free not available Number not yet purchased, ported, or activated Confirm procurement or porting status with carrier

Quick Reference

Question Answer
Where do you manage DID and toll-free numbers? Admin → Telephony → DID Numbers
What are the two main tabs? DID Ranges and DID Assignments
What can a number be assigned to? A person, a phone, or a call flow
What fields are needed to create a range? DID Start, DID End, and Service Provider
Can toll-free numbers be managed here too? Yes — same workflow
What must happen before a number can be assigned? It must exist in a DID Range

Naming Convention

Resource Example
DID Range (provider) CarrierA_US_DID_Block_01
Toll-Free main entry US_TF_Main_Inbound

See Also

Screenshots

To unassign

DID Ranges

Edges & Edge Groups

Navigation: Admin → Telephony → Edges / Admin → Telephony → Edge Groups Last verified: Genesys Cloud Resource Center — March 2026

What Are Edges?

An Edge is a BYOC Premises network appliance that handles local media and provides telephony services including media server, SIP registrar, and SIP proxy functions. Edges are the core infrastructure component of BYOC Premises deployments.

ℹ️ Edges and Edge Groups are a BYOC Premises concept. They do not apply to BYOC Cloud or Genesys Cloud Voice deployments.

What Are Edge Groups?

An Edge Group is a set of BYOC Premises Edges directly connected over a high-bandwidth, low-latency network (LAN or WAN). Edges in the same group can share trunks and related telephony resources with each other.

Resource Types That Can Be Shared Examples
Phone trunks SIP phone trunks
Communication provider trunks Carrier SIP trunks
External gateways SBC/gateway resources
SIP carriers and VoIP gateways Shared across grouped Edges

⚠️ Different Edge Groups do not share resources with each other. Only group Edges that are on a suitably low-latency, high-bandwidth link.

Navigation

Task Path
Open Edges Admin → Telephony → Edges
Provision new Edge Edges → Provision New Edge
View Edge details Edges → select Edge → information panel
Open Edge Groups Admin → Telephony → Edge Groups
Create Edge Group Edge Groups → Create

Provisioning an Edge

Step Action
Step 1 Navigate to Admin → Telephony → Edges
Step 2 Click Provision New Edge
Step 3 Enter Edge Name
Step 4 Select the hardware solution type (e.g. BYOC Premises – Customer Hardware Solution)
Step 5 Enter Serial Number and confirm it
Step 6 Click Provision Edge
Step 7 Configure the Edge's network interface(s)
Step 8 Associate with the correct Site and Edge Group

Edge Provisioning Fields

Field Description
Edge Name Identifier for the Edge
Hardware Solution Type Selects the Edge model/solution (e.g. Customer Hardware Solution)
Serial Number Physical hardware serial number
Confirm Serial Number Confirmation field to prevent entry errors

Edge Information Panel

After provisioning, the Edge information panel shows:

Field Description
Connectivity status Cloud connectivity and operational state
Trunk status Associated trunk state
Software version Installed/staged version
Hardware model Edge hardware model
Serial number Hardware serial number
Pairing ID Used during provisioning
Metrics Call capacity and CPS visibility

Creating an Edge Group

Step Action
Step 1 Navigate to Admin → Telephony → Edge Groups
Step 2 Click Create
Step 3 Enter the Edge Group Name
Step 4 Add one or more Edges to the group
Step 5 Associate trunk(s) as needed
Step 6 Save

ℹ️ Plan sites and trunks before creating Edge Groups. Genesys recommends determining required trunks and sites first.

Redundancy

Genesys recommends N+1 redundancy for BYOC Premises Edges. Managed phones register with both a primary and secondary Edge. If the primary Edge becomes unavailable, phones switch to the secondary — though UI lag of up to 15 seconds may occur during the transition.

For proper load distribution, keep Edge call capacities similar within the same design.

Edge Security

Security Feature Description
Mutual TLS Edge control communications use mTLS/HTTPS to Genesys Cloud
Outbound-only connections Edges initiate connections outbound — no need to expose the Edge directly on the internet
CA trust Related to Certificate Authorities configuration for remote TLS endpoints

Key Design Rules

Rule Detail
Network requirement Edge Groups require high-bandwidth, low-latency LAN or WAN between grouped Edges
Cross-group isolation Different Edge Groups do not share resources
Build order Create sites and plan trunks before grouping Edges
Capacity Keep Edge capacities similar for predictable load distribution and failover

Troubleshooting

Issue Cause Resolution
Edge offline / unavailable Network, pairing, software, or service issue Check Edge information panel, connectivity, software version, and network path
Trunks not shared across Edges Edges not in same Edge Group or network latency too high Verify Edge Group membership and low-latency connectivity
Phones fail over unexpectedly Primary Edge unavailable Validate primary/secondary Edge design and registration behavior
Calls fail after update Edge software change or maintenance timing issue Review staged/installed version; schedule updates with call draining
Edge not provisioning Incorrect hardware type or serial entry Verify hardware type and serial number before reprovisioning

Quick Reference

Question Answer
What is an Edge? A BYOC Premises media appliance — media server, SIP registrar, and SIP proxy
What is an Edge Group? A set of BYOC Premises Edges on a high-bandwidth, low-latency network that share trunks and resources
What fields are used to provision an Edge? Name and serial number
Why use Edge Groups? To share trunks/resources and support local routing and resiliency
What redundancy model does Genesys recommend? N+1
Do different Edge Groups share resources? No

Naming Convention

Resource Example
Edge MTY_Edge_01
Edge MTY_Edge_02
Edge Group MTY_Core_Group
Customer Hardware Edge MTY_CHS_Edge_01

Pattern: <Location>_<ObjectType>_<Sequence>

See Also

Screenshots

Extensions

Navigation: Admin → Telephony → Extensions Last verified: Genesys Cloud Resource Center — March 2026

What Are Extensions?

Extensions are internal dialing numbers that allow users to reach each other within the organization without using a full DID number. Before an extension can be assigned to a user, it must first exist in an Extension Pool.

Two Main Areas

Tab Purpose
Extension Pools Create and manage the inventory of available extension numbers
Assignments Search and review how extensions are currently assigned to users

The Pool-First Model

Create Extension Pool
        ↓
Assign Pool to a Division
        ↓
Extensions become available for assignment
        ↓
Assign extension to user via Contact Information

⚠️ You cannot assign an extension to a user until it exists in an Extension Pool.

Navigation

Task Path
Open Extensions Admin → Telephony → Extensions
Open Extension Pools Extensions → Extension Pools tab
Open Assignments Extensions → Assignments tab
Add extension(s) Extension Pools → Add
Assign to user Admin → People and Permissions → People → [User] → Contact Information

Required permissions:

Step 1: Create an Extension Pool

Step Action
Step 1 Navigate to Admin → Telephony → Extensions
Step 2 Open the Extension Pools tab
Step 3 Click Add
Step 4 Enter Extension Start (single number, or first in a range)
Step 5 Enter Extension End — leave blank for a single extension, or fill for a range
Step 6 Select the Division
Step 7 Click Create

Extension Pool Fields

Field Description
Extension Start Single extension number, or first number in a range
Extension End Last number in a range; leave blank for a single extension
Division Access-control boundary for this pool

Step 2: Assign Extension to a User

Extensions are assigned through the user's profile, not from the Extensions page directly.

Step Action
Step 1 Navigate to Admin → People and Permissions → People
Step 2 Search for the user and open Edit Person
Step 3 Open the Person Details tab
Step 4 Click View Edit Mode
Step 5 In Contact Information, click Edit
Step 6 Under Phone, enter the extension in the ext. field of an empty Work Phone entry
Step 7 Click Save
Step 8 Return to Extensions → Assignments to confirm the extension appears

⚠️ Critical: Enter the extension only in a Work Phone entry that does not already have a phone number. Adding an extension to an entry that already contains a number prevents Genesys Cloud from generating a dial plan for the user, and the extension will not appear in the Assignments page.

Key Rules

Rule Detail
Pool first Extension must exist in a pool before it can be assigned
Unique extensions Duplicate extension numbers are not allowed
Empty Work Phone entry Extension must be entered in an empty Work Phone slot, not alongside an existing number
DID and extension are separate If a user has both, they must be separate phone entries
Range deletion If an extension was added as part of a range, the entire range may need to be deleted — you cannot delete a single number from within a range
Assigned range deletion Cannot delete an extension pool while any extension in its range is assigned to a user
Propagation delay After assigning an extension, it can take up to 60 minutes before it is accessible in a Dial By Extension action

Assignments View

The Assignments tab lets you:

Troubleshooting

Issue Cause Resolution
Extension not assignable Not added to an Extension Pool Add it to Extension Pools first
Duplicate extension error Same number already exists Use a unique extension
Extension not visible in Assignments Entered in a Work Phone entry that already had a number Remove and re-enter in an empty Work Phone entry
Cannot delete extension Belongs to a range or is currently assigned Remove the user assignment first, or delete the entire range
User not receiving calls by extension Profile or dial-plan issue Verify pool membership, user profile entry, and permissions
Dial By Extension not working immediately Propagation delay Wait up to 60 minutes and retest

Quick Reference

Question Answer
Where do you manage extensions? Admin → Telephony → Extensions
What are the two main areas? Extension Pools and Assignments
What must happen before assigning an extension? It must be added to an Extension Pool
Can duplicate extensions exist? No
Where is the extension assigned to a user? In the user's Contact Information, in the ext. field
Can you delete one number from a range? Not if it was originally entered as part of a range
How long can Dial By Extension take to recognize a new extension? Up to 60 minutes

Naming Convention

Resource Example
Extension Pool Support_Ext_Pool_4100_4199
Extension Pool Sales_Ext_Pool_4200_4299

Pattern: <Division>_Ext_Pool_<Start>_<End>

See Also

Screenshots

Sites

Navigation: Admin → Telephony → Sites Last verified: Genesys Cloud Resource Center — March 2026

What Is a Site?

A site is the home of a set of phones. It defines the telephony dialing properties, call classification rules, and outbound routing rules for the phones assigned to it. Every phone in Genesys Cloud belongs to a site, and the site determines where calls go and how numbers are interpreted.

Sites are used across all telephony deployment models: BYOC Cloud, Genesys Cloud Voice, and BYOC Premises.

⚠️ The Media Model (Cloud or Premises) cannot be changed after site creation. Choose carefully.

Site Tabs

Tab Purpose
General Description, default site toggle, media regions, Relay/TURN behavior, outbound caller ID
Number Plans Classify and normalize dialed numbers; default plans provided; max 200 per site
Outbound Routes Route calls to external trunks; Sequential or Random distribution
Simulate Call Validate routing configuration without placing an actual call

Navigation

Task Path
Open sites Admin → Telephony → Sites
Create site Sites → Create New
Configure General settings Sites → [Site] → General
Add number plans Sites → [Site] → Number Plans
Create outbound route Sites → [Site] → Outbound Routes
Run call simulator Sites → [Site] → Simulate Call

Creating a Site — Field Reference

Create Form

Field Description Notes
Site Name Unique name for the site Required
Location Location assigned to the site Only locations marked as available for sites appear; required
Time Zone Time zone for the site Required
Media Model Cloud or Premises Cannot be changed after creation

General Tab

Field Description Notes
Description Free-text description Optional
Make this Site my default Site Sets org-wide default Only one default site allowed
Media Regions Select media regions for WebRTC / Global Media Fabric Relevant for WebRTC deployments
Relay/TURN Behavior Controls TURN relay region selection for WebRTC calls Two options: Any media region set on this site / Lowest latency via Geo-Lookup
Caller Address Outbound caller number Must be in E.164 format
Caller Name Outbound caller name Text

Number Plans Tab

Genesys provides default number plans. Custom plans can be added to control what users can dial and how numbers are normalized before route selection. Maximum of 200 number plans per site.

Outbound Routes Tab

Field Description
Route Name Name for the outbound route
Classification Number plan classification this route handles
External Trunk(s) One or more trunks the route uses
Distribution Sequential (ordered failover) or Random (load distribution)
State Enable/disable the route

Simulate Call Tab

Enter a destination number or SIP address and click Simulate. The tool validates:

ℹ️ Simulate Call validates configuration but does not place an actual call.

Media Model Selection

Deployment Media Model
BYOC Cloud Cloud
Genesys Cloud Voice Cloud
BYOC Premises Premises

Step-by-Step: Create a Site

Step Action
Step 1 Navigate to Admin → Telephony → Sites
Step 2 Click Create New
Step 3 Enter Site Name, select Location, Time Zone, and Media Model
Step 4 Click Create Site
Step 5 On General, add description and optionally set as default site
Step 6 Configure Caller Address (E.164) and Caller Name
Step 7 Configure Media Regions and Relay/TURN Behavior if using WebRTC
Step 8 Add number plans on Number Plans tab
Step 9 Create one or more outbound routes on Outbound Routes tab
Step 10 Run Simulate Call to validate routing before going live

Key Constraints

Constraint Detail
Media Model Cannot be changed after creation
Location availability Only locations marked available for sites appear in the selector
Default site Only one site can be the default
Number plans per site Maximum 200
Caller Address format Must be E.164 (e.g. +528181234567)

Troubleshooting

Issue Cause Resolution
Location not visible in selector Location not marked as available for sites Update the location setting
Caller ID not displaying correctly Caller Address not in E.164 or overridden by Prioritized Caller Selection Recheck format and caller selection config
Calls not routing Number plan or outbound route mismatch Use Simulate Call to trace normalization, classification, and route selection
No route selected Route disabled or no matching classification Verify route state, classification, and selected trunks
Simulator shows failure Site, trunk, Edge, or destination settings incomplete Review each simulator output field in order

Quick Reference

Question Answer
What is a Site? The home of a set of phones; defines classification, routing, and dialing rules
What are the four tabs? General, Number Plans, Outbound Routes, Simulate Call
What media models exist? Cloud and Premises
Can you change the media model later? No
What does Simulate Call do? Validates routing config without placing a real call
What distribution patterns exist for outbound routes? Sequential and Random
What format must Caller Address use? E.164

Naming Convention

Resource Example
Cloud site MTY_Main_Cloud
Premises site MTY_Branch_Prem
Outbound route PSTN_Main
Number plan MX_National_Dialing

Pattern: <Location>_<Purpose>_<MediaModel>

See Also

Screenshots

Topology

Navigation: Admin → Telephony → Topology Last verified: Genesys Cloud Resource Center — March 2026

What Is Topology?

Topology is the administrative map view of the Genesys Cloud telephony network. It displays how telephony components relate to each other — sites, phones, edges, and trunks — and is especially useful for troubleshooting offline or out-of-service edges and validating phone-to-edge assignments.

ℹ️ Topology is a monitoring and diagnostic tool, not a provisioning wizard. Changes to telephony objects are made in their respective admin pages; Topology is where you confirm the result.

Navigation

Task Path
Open Topology Admin → Telephony → Topology
Drill into an object Click any site, edge, trunk, or phone in the map
Troubleshoot edges Select the edge object to view status details

What Topology Shows

Object Description
Sites Logical telephony routing entities
Edges Media-handling devices in the telephony environment
Trunks Carrier or PBX connectivity into Genesys Cloud
Phones Endpoints and their edge/site assignments
Phone-to-Edge Assignments Shows which phones connect to which edges, including primary and secondary site relationships
Status Indicators Visual state of objects — helps identify offline or out-of-service edges

How to Use Topology

Step Action
Step 1 Navigate to Admin → Telephony → Topology
Step 2 Review the map for your organization's telephony objects
Step 3 Look for offline or out-of-service edge indicators
Step 4 Click into a suspect object to drill down into its details
Step 5 Enable phone-to-edge assignment view to validate primary/secondary site relationships
Step 6 Use findings to continue troubleshooting in the relevant admin page (Sites, Trunks, Edges)

Key Use Cases

Scenario Description
Incident triage Quickly visualize where a telephony problem exists before diving into logs
Post-change validation Confirm expected object relationships after site, trunk, or edge changes
Resiliency review Validate phone-to-edge assignments and primary/secondary site design
Edge troubleshooting Identify offline or out-of-service edges at a glance
Onboarding review Confirm a new telephony deployment is connected as designed

Troubleshooting

Issue Cause Resolution
Edge appears offline Edge, network, or service issue Drill into the edge; continue in Admin → Telephony → Edges
Phones not where expected Assignment or site relationship misconfiguration Review phone-to-edge assignments and site design
Map looks incomplete Telephony objects not yet configured or not in expected state Verify sites, trunks, edges, and phones exist and are correctly assigned
Trunk/site relationship unclear Naming inconsistency or design ambiguity Standardize naming; compare with site/trunk config pages

Best Practices

Practice Reason
Review Topology after major telephony changes Visually confirms that relationships updated as expected
Use Topology early when troubleshooting Narrows the fault domain before deeper investigation
Validate phone-to-edge assignments regularly Prevents unnoticed resiliency or registration issues
Keep site and trunk naming consistent Makes the map easier to read and interpret
Use Topology for visibility; use admin pages for fixes Topology shows the problem — the fix happens elsewhere

Quick Reference

Question Answer
What does Topology show? Sites, phones, edges, and trunks in a visual map
What is it mainly used for? Visualization and troubleshooting
Can you make changes in Topology? No — it is read-only for monitoring and diagnostics
What edge issue does it help with? Identifying offline or out-of-service edges
Does it show resiliency design? Yes — phone-to-edge assignments show primary/secondary site relationships

See Also

Screenshots

Trunks

Navigation: Admin → Telephony → Trunks (or Admin → Telephony → BYOC Cloud → Trunks) Last verified: Genesys Cloud Resource Center — March 2026

What Are Trunks?

A trunk is the SIP communications link between Genesys Cloud and an external carrier or PBX. Trunks carry inbound and outbound SIP signaling and are consumed by Sites via Outbound Routes to send calls to the PSTN or connected telephony infrastructure.

Trunk Types

Trunk Type Deployment Model Used For
BYOC Carrier BYOC Cloud Third-party SIP carrier connectivity over the public internet
BYOC PBX BYOC Cloud SIP interconnect with an existing IP-PBX over the public internet
External SIP BYOC Premises On-premises SIP carrier or PBX connectivity through an Edge

ℹ️ BYOC Carrier and BYOC PBX are for BYOC Cloud. External SIP is for BYOC Premises. Do not mix deployment models.

Navigation

Task Path
Open trunks Admin → Telephony → Trunks
Open BYOC Cloud trunks Admin → Telephony → BYOC Cloud → Trunks
Create a trunk Trunks → Create Trunk
Edit a trunk Trunks → select trunk → Edit
Use trunk in routing Admin → Telephony → Sites → [Site] → Outbound Routes

Creating a BYOC Carrier Trunk — Field Reference

Field Description Notes
Name Trunk name Use a descriptive, consistent naming convention
Type BYOC Carrier / BYOC PBX / External SIP Determined by your deployment model
Subtype Vendor/carrier profile where applicable Optional
State Operational state Set to In Service when ready for production
Transport Protocol SIP transport used to send calls UDP / TCP / TLS — does not control inbound protocol
Inbound SIP Termination Identifier Regionally unique ID for inbound SIP routing Required for BYOC Carrier; confirm with carrier
Outbound Request URI Controls SIP request routing for outbound calls Carrier-specific
SIP Servers / Proxies Remote SIP server or proxy addresses Carrier-provided
Digest Authentication SIP authentication Enable if required by the carrier or PBX
Caller Address / Caller ID Outbound caller number E.164 format
Caller Name Outbound caller name Text
SIP Access Control IP allowlist for inbound SIP signaling Restrict to carrier signaling IPs only
PBX Passthrough Enables PBX passthrough where supported Optional
Custom SIP Headers Additional SIP header configuration Optional

Transport Protocol Behaviour

Protocol Notes
UDP Standard, connectionless — widely supported
TCP Connection-oriented, more reliable for SIP
TLS Encrypted SIP signaling; pairs with SRTP for full call security

⚠️ For BYOC Cloud, the transport protocol setting controls how Genesys sends calls on the trunk. It is not enforced on calls received on that trunk.

Step-by-Step: Create a BYOC Carrier Trunk

Step Action
Step 1 Navigate to Admin → Telephony → BYOC Cloud → Trunks
Step 2 Click Create Trunk
Step 3 Select BYOC Carrier as the trunk type
Step 4 Enter the trunk Name
Step 5 Set State to In Service
Step 6 Select Transport Protocol
Step 7 Enter the Inbound SIP Termination Identifier
Step 8 Configure Outbound Request URI
Step 9 Enter SIP Servers / Proxies
Step 10 Enable Digest Authentication if required
Step 11 Under Calling, set Caller ID and Caller Name
Step 12 Configure SIP Access Control IP rules
Step 13 Save the trunk
Step 14 Add the trunk to a Site → Outbound Route
Step 15 Validate with test calls or Simulate Call

Dependencies

Component Purpose
Sites Outbound routes on sites reference external trunks
Number Plans Classify dialed numbers before route/trunk selection
Outbound Routes Select one or more trunks with Sequential or Random distribution
Carrier / PBX Remote SIP endpoint the trunk connects to
Certificate Authorities Required when using TLS trunks (BYOC Premises)

Troubleshooting

Issue Cause Resolution
Trunk not sending calls Wrong transport protocol or routing config Recheck protocol, URI, and remote endpoint requirements
Inbound calls fail Incorrect inbound SIP identifier Validate inbound SIP termination identifier with carrier
Secure calls fail TLS/certificate mismatch Validate TLS support and certificate/trust configuration
Unauthorized SIP traffic SIP ACL not configured Restrict signaling IPs using SIP Access Control
Wrong outbound identity Caller ID/name misconfigured Recheck Calling section values
Route not selecting trunk Number plan or outbound route misconfiguration Validate number plans, route classification, trunk selection

Quick Reference

Question Answer
What trunk types exist? BYOC Carrier, BYOC PBX, External SIP
Which are for BYOC Cloud? BYOC Carrier and BYOC PBX
Which is for BYOC Premises? External SIP
What transport protocols are supported? UDP, TCP, TLS
What does SIP Access Control do? Permits signaling only from specific IP addresses
What is the Inbound SIP Termination Identifier? A regionally unique ID used for inbound SIP routing on BYOC Carrier

Naming Convention

Resource Example
Carrier trunk CarrierA_BYOCCarrier_Prod
PBX trunk CorpPBX_BYOCPBX_Test
Premises SIP trunk HQ_ExternalSIP_Primary

Pattern: <Provider>_<TrunkType>_<Environment>

See Also

Screenshots

WebRTC Phone Management

Navigation: Admin → Telephony → Phone Management Last verified: Genesys Cloud Resource Center — March 2026

What Is a WebRTC Phone?

A Genesys Cloud WebRTC phone is a browser or desktop app-based softphone that lets users place and receive calls directly in the Genesys Cloud client — no physical desk phone required. It is the most common phone type for contact center agents, remote users, and fast deployments.

Provisioning Model

WebRTC phones are provisioned in two steps:

Step 1: Create Base Settings
        ↓
Step 2: Create the Phone object

Base Settings is a shared configuration profile that defines how the WebRTC phone behaves. The Phone is the individual user-assigned record that uses those settings. Always build Base Settings first.

Navigation

Task Path
Open Phone Management Admin → Telephony → Phone Management
Create WebRTC Base Settings Phone Management → Base Settings tab → Add
Create WebRTC Phone Phone Management → Phones tab → Create New
Configure global WebRTC behavior Admin → Telephony → Global Telephony Settings
Configure site media behavior Admin → Telephony → Sites → [Site] → General
User selects WebRTC phone Genesys Cloud client → Calls panel → phone selector

Required permission: Telephony > Plugin > All

Step 1: Create Base Settings

Step Action
Step 1 Navigate to Admin → Telephony → Phone Management
Step 2 Open the Base Settings tab
Step 3 Click Add
Step 4 Enter a Base Settings Name
Step 5 In Phone Make and Model, select Genesys Cloud WebRTC Phone
Step 6 Enable Persistent Connection if needed
Step 7 Configure Transport DSCP Value and Media DSCP Value
Step 8 Click Save Base Settings

Base Settings Fields

Field Description Notes
Base Settings Name Name for this configuration profile Use a descriptive name e.g. Support_WebRTC_Standard
Phone Make and Model Select the phone type Choose Genesys Cloud WebRTC Phone
Persistent Connection Keeps the WebRTC connection open after calls end Improves subsequent call handling speed
Persistent Connection Timeout How long the connection stays active Configure based on call volume patterns
Transport DSCP Value QoS marking for SIP signaling traffic Align with enterprise voice network policy
Media DSCP Value QoS marking for audio/media traffic Align with enterprise voice network policy

Step 2: Create the Phone

Step Action
Step 1 In Phone Management, open the Phones tab
Step 2 Click Create New
Step 3 Enter the Phone Name
Step 4 Select the Site
Step 5 Select the Base Settings profile created in Step 1
Step 6 Assign the User
Step 7 Click Save
Step 8 Have the user select the WebRTC phone in the client and test calling

Persistent Connection

Keeping the WebRTC connection open after a call ends allows subsequent calls to alert faster because the connection is already established.

Setting Behaviour
Disabled Connection closes after each call; next call requires fresh connection setup
Enabled Connection stays open for the timeout period; subsequent calls alert immediately

⚠️ Important: If you enable Persistent Connection after users are already logged in, they must log out and back in for the setting to apply. Genesys recommends making this change outside business hours.

QoS / DSCP

DSCP values mark WebRTC SIP and media traffic so your network can prioritize it. Set these values to match your organization's enterprise voice QoS policy.

DSCP Field Applies To
Transport DSCP SIP signaling traffic
Media DSCP Audio/RTP media traffic

Site-Level WebRTC Media Settings

These are configured on the Site, not in Base Settings. Validate these for every site that will host WebRTC users.

Field Options Description
Media Regions Available Home/Core/Satellite regions Select and prioritize regions for WebRTC / Global Media Fabric
Relay/TURN Behavior Any media region set on this site / Lowest latency via Geo-Lookup Controls how Genesys selects TURN relay regions for WebRTC calls that need relay services
Relay/TURN Option Best For
Any media region set on this site Strict control — limits TURN relay to configured regions only
Lowest latency via Geo-Lookup Best performance — Genesys dynamically selects lowest-latency TURN region

⚠️ Forcing TURN relay can reduce resiliency and force RTP through relay services when not otherwise necessary.

User Experience

Users select and manage the WebRTC phone from the Calls panel in the Genesys Cloud client:

ℹ️ Recommend using a quality headset rather than laptop built-in speakers and microphone to avoid echo and audio quality issues.

Troubleshooting

Issue Cause Resolution
User cannot answer calls reliably Persistent connection disabled or not applied Enable it; have user log out and back in
No audio Wrong microphone/speaker selected Verify audio devices in WebRTC phone client settings
Poor call quality DSCP/network/headset issue Check QoS policy, headset, and local network
WebRTC phone not visible to user Phone not created or not assigned to correct user Recheck phone assignment
Calls do not route Site/routing configuration issue Validate site, number plans, and trunks
Settings change not applied User session retained old settings Log out and back in
Unexpected TURN/media path Site Media Regions or Relay/TURN Behavior misconfigured Review assigned Site's General tab

Quick Reference

Question Answer
How do you add a WebRTC phone? Create Base Settings first, then create the Phone
What model do you select? Genesys Cloud WebRTC Phone
Why enable Persistent Connection? Improves subsequent call handling speed
Where is Relay/TURN Behavior configured? On the Site, not in Base Settings
What should users do after Persistent Connection is enabled later? Log out and back in

Naming Convention

Resource Example
Base Settings Support_WebRTC_Standard
Base Settings Sales_WebRTC_Remote
Phone AgentName_WebRTC

Pattern: <BusinessArea>_WebRTC_<Purpose>

See Also

Screenshots

Now Add a phone

Phone Management

Section Description
Feature Area Telephony Infrastructure
Navigation Admin → Telephony → Phone Management
Alt Navigation Menu → Digital and Telephony → Telephony → Phone Management
Primary Function Configure, provision, and manage the physical and software phones used by agents to make and receive calls

Genesys Cloud supports multiple phone types. Phone Management is where administrators create phone records, assign base settings, connect phones to sites, and manage provisioning.

Study Notes

Topic Explanation
Phone Management Admin area for creating and managing phone configurations — assigns phones to users and sites
Base Settings A reusable configuration profile applied to phones — defines codec, DTMF method, TLS settings, Quality of Service, and more
Site A logical grouping of telephony resources (trunks, number plans, outbound routes) — phones are assigned to a site
Provisioning The process of automatically delivering a phone's configuration from Genesys Cloud to the physical device
Zero Touch Provisioning (ZTP) Managed phones can self-configure by contacting the Genesys provisioning server on first boot
Hardware ID The identifier used to associate a phone record with a physical device (e.g., MAC address for hardware phones, FQDN for softphones)
Line Keys Configurable buttons on a hardware phone — can be assigned to speed dial, BLF (Busy Lamp Field), or line registrations
HELD HTTP-Enabled Location Delivery — protocol allowing Poly phones to retrieve precise location from a LIS server for E911 accuracy

Four Phone Categories

Category Description Configuration Use Case
Managed Genesys Cloud controls the full configuration — provisioned via HTTPS with TLS and redundancy Configured entirely in Genesys Cloud via Phone Management and Base Settings Hardware desk phones (Poly VVX, Poly Edge E, AudioCodes)
Unmanaged Phone registers with Genesys Cloud but is configured externally Only basic SIP connection info in Genesys Cloud; uses a generic SIP base settings profile Any SIP-compliant device; FXS analog adapters
WebRTC Browser-based softphone — no hardware or separate software required Enabled per-user; headset connected to computer Agents working in browser; remote workers
Remote An external phone number or SIP address (e.g., cell phone) Configured as a remote number — calls are bridged to the remote device when an interaction is accepted Mobile workers; home phones; non-networked devices

Managed vs Unmanaged — Key Differences

Attribute Managed Unmanaged
Configuration source Genesys Cloud (provisioned via HTTPS) External to Genesys Cloud
Default base settings profiles Yes — Genesys provides model-specific defaults No — uses generic SIP profile
TLS / SRTP Automatic via provisioning Possible but requires manual configuration
Redundancy (primary + secondary SIP) Automatic Possible but not automatic
Mutual authentication (Genesys Cloud Voice) Standard Not supported
ZTP support Yes No
Examples Poly Edge E, Poly VVX, AudioCodes Generic SIP devices, FXS adapters

WebRTC Phones

Attribute Detail
No hardware required Runs entirely in the browser
No software download Built into Genesys Cloud web app
Setup Enable WebRTC phone per user; connect a headset
Creates a dedicated phone line Provisioning a WebRTC phone creates a specific phone line for that user
Recommended for Remote workers, browser-first environments

Remote Phones

Attribute Detail
What it is An external phone number or SIP address used to connect a user to Genesys Cloud calls
How it works When a call is placed/answered in the browser, Genesys Cloud calls the remote number to bridge the connection
Routing Follows the site's numbering plans and outbound routes
Typical use Mobile workers, agents who use personal cell phones

Navigation and Configuration

Task Path
Open Phone Management Admin → Telephony → Phone Management
View/manage phones Phone Management → Phones tab
View/manage base settings Phone Management → Base Settings tab
Add a phone Phone Management → Phones → Add Phone
Assign base settings to a phone Select phone → choose Base Settings profile
Assign phone to a site Select phone → choose Site
Restart a phone Phone Management → select phone → Restart
Log out a phone Phone Management → select phone → Log Out
Migrate base settings Phone Management → select phones → Migrate Base Settings

Base Settings

Base Settings are reusable configuration templates applied to one or more phones of the same model. They define:

Setting Category Examples
General Dynamic Reload (auto-updates without manual restart)
Media / Quality of Service DSCP value for RTP packets; RTP Audio Port Start Range (default 4000; range 1024–65,535)
Codecs Preferred codec list (MIME format, priority-ordered)
DTMF RTP Events (out-of-band, RFC 4733 — default) or In-band Audio; payload type 96–127 for RTP Events
Security TLS certificate authority selection; mutual authentication
Provisioning Firmware version; firmware update settings
E911 / HELD Enable HELD; provide LIS URL and Emergency Routing Service Account ID (for Poly VVX, CCX, Edge E)

Managed Phone Provisioning Process

Phone powers on
        ↓
Phone contacts Genesys Cloud provisioning server (HTTPS)
        ↓
Genesys Cloud delivers configuration (base settings, line keys, TLS certs, SIP credentials)
        ↓
Phone registers with Genesys Cloud SIP infrastructure
        ↓
Phone is ready — appears as Online in Phone Management

Supported Managed Phone Brands (Examples)

Brand Example Models
Poly (formerly Polycom) Poly Edge E Series (E100/E220/E300/E320/E350/E400/E450/E500/E550) · VVX Series · SoundPoint IP · SoundStation
AudioCodes Various models (note: some AudioCodes models are incompatible with Genesys Cloud Voice/BYOC Cloud)
Spectralink 84-Series (incompatible with Genesys Cloud Voice/BYOC Cloud)

Compatibility note: Most managed phones are compatible with Genesys Cloud Voice and BYOC Cloud. Some AudioCodes models and certain older SoundPoint models are incompatible — refer to the Genesys Cloud Voice / BYOC Cloud compatible phones matrix before purchasing.

Phone Management Operations

Operation Description
Add Phone Create a new phone record; assign name, base settings, site, and hardware ID
Import Phones Bulk import via CSV
Restart Phone Sends a restart command to the managed phone over the network
Log Out Phone Logs the user off the phone remotely
Migrate Base Settings Move phones to a different base settings profile (e.g., after a model upgrade)
Edit Phone Change name, base settings, site assignment, line key configuration
Check Status View online/offline status for each phone
Filter Filter phone list by site, base settings, status, or name

Permissions

Permission Purpose
Telephony > Plugin > All Full access to telephony admin including Phone Management
Telephony > SitesManagedPhones > View/Add/Edit/Delete Phone-specific permissions

Key Takeaways

Topic Summary
Phone categories Managed · Unmanaged · WebRTC · Remote
Managed phones Fully provisioned by Genesys Cloud via HTTPS — TLS, redundancy, ZTP automatic
Unmanaged phones Configured externally; only SIP registration info in Genesys Cloud; uses generic profile
WebRTC Browser-based; no hardware; headset required; per-user enablement
Remote External number (cell phone, SIP address) bridged to calls
Base Settings Reusable profile — codec, DTMF, TLS, QoS, firmware, HELD
Navigation Admin → Telephony → Phone Management
Phones tab Manage individual phone records
Base Settings tab Manage configuration templates

E911 and Emergency Locations

Section Description
Feature Area Telephony Infrastructure
Navigation (Sites / Number Plans) Admin → Telephony → Sites → [site] → Number Plans tab
Navigation (E911 Kari's Law) Contact Genesys Cloud Voice support directly to configure Kari's Law notifications
Navigation (HELD for Poly phones) Admin → Telephony → Trunks → External Trunks → [trunk] → General → Outbound → Location Conveyance
Navigation (Location Details) Admin → Telephony → Locations (for physical address configuration)
Primary Function Route emergency calls (911) to the correct Public Safety Answering Point (PSAP) based on the caller's location, and comply with Kari's Law requirements

Study Notes

Topic Explanation
E911 Enhanced 911 — automatically transmits the caller's address and telephone number to the emergency dispatcher when 911 is dialed; no need for the caller to state their location
Traditional 911 Caller must identify their location manually
PSAP Public Safety Answering Point — the regional emergency services dispatch center that receives 911 calls
Kari's Law US federal law (effective February 16, 2018) — requires multi-line telephone systems (MLTS) to: (1) allow direct 911 dialing without a prefix, and (2) send a notification to a designated person when 911 is dialed
MLTS Multi-Line Telephone System — any phone system with multiple lines; contact centers are MLTS operators
Location Details Configuration in Genesys Cloud that stores a physical location address — used for E911 routing and Kari's Law notifications
HELD HTTP-Enabled Location Delivery — protocol that allows Poly phones to query a Location Information Server (LIS) for their precise network-based location at call time
LIS Location Information Service — a server that maps network location data (IP, MAC) to a civic address for E911 purposes

Kari's Law Requirements (US Only)

Kari's Law applies to Genesys Cloud Voice customers in the United States.

Requirement Genesys Cloud Voice Behavior
Direct 911 dialing (no prefix) Automatically satisfied — no customer action required
Notification to designated location when 911 is dialed Requires configuration — must be set up with Genesys Cloud Voice support

How to Configure Kari's Law Compliance (Genesys Cloud Voice)

  1. Contact Genesys Cloud Voice support
  2. Provide the following:
    • Full location address (US addresses only — Canadian addresses do not support notification)
    • Email addresses or email-as-text addresses to notify when 911 is dialed (e.g., user@company.com, 5551234567@txt.att.net)

Notification is triggered based on the physical location address configured in Location Details.

Configuring Emergency Numbers in Sites

For all telephony options, emergency numbers are configured at the site level.

Step Path
Open Admin Admin → Telephony → Sites
Select site Choose the appropriate site
Open Number Plans tab Click Number Plans
Select Emergency plan Click on the Emergency number plan in the list
Enter emergency number Type the emergency services number (e.g., 911 for the US)
Kari's Law note US users must not alter the 911 number with a prefix or any other modification

Warning: Do not assign an emergency number plan to a BYOC trunk unless you have verified with your carrier that they provide emergency services and that the carrier has the correct location for your phone numbers.

BYOC and Emergency Services

Genesys Cloud Voice includes built-in E911 support. BYOC customers must arrange E911 separately:

Option E911 Approach
Genesys Cloud Voice E911 included — configured through Location Details and Kari's Law setup with Genesys support
BYOC Cloud Must check with your carrier — carrier must support E911 for your numbers and locations
BYOC Premises Must check with your carrier — same requirement; also need to verify site-level number plan configuration

For BYOC E911 setup: Admin → Telephony → Trunks → BYOC trunk → configure as directed by your carrier

Reference article: "Set up emergency services with BYOC" in the Genesys Cloud Resource Center.

HELD — HTTP-Enabled Location Delivery (Poly Phones)

HELD allows Poly phones to retrieve their precise network location from a LIS server and include it in the SIP INVITE when a 911 call is placed, enabling more accurate emergency routing.

Supported phones: Poly VVX, Poly CCX, Poly Edge E

HELD Configuration Steps

Step Where
1. Enable Location Conveyance on trunk Admin → Telephony → Trunks → External Trunks → [trunk] → General → Outbound → check Location Conveyance
2. Enter Emergency Routing Service Account ID From your emergency service provider (or token ID if token authentication is required)
3. Enter Location Information Server URL URL to send HELD requests to
4. Enable HELD in phone Base Settings Admin → Telephony → Phone Management → Base Settings → [Poly base settings] → enable HELD

How E911 Works — Genesys Cloud Voice

Agent dials 911
        ↓
Genesys Cloud Voice looks up the physical location address associated with the agent's number / location
        ↓
Call routed to appropriate PSAP for that address
        ↓
PSAP receives caller's address and telephone number automatically (E911)
        ↓
Kari's Law notification sent to designated email/SMS addresses

Locations Configuration

Physical location addresses are configured in: Admin → Telephony → Locations

Each location stores:

E911 for Remote Workers

Remote workers present a challenge because their physical location is not fixed. Genesys Cloud Voice provides E911 configuration options for remote workers — the physical location address must be kept current for accurate PSAP routing.

Consideration Detail
Accurate location data required Inaccurate location may route the 911 call to the wrong PSAP — potentially causing delays
National fallback If E911 cannot locate the caller, the call may route to a national emergency response service (less accurate, slower)
Remote workers Must have their location updated when they change physical locations

Key Takeaways

Topic Summary
E911 vs 911 E911 automatically transmits caller address and number to PSAP; traditional 911 requires caller to state location
Kari's Law US federal law — MLTS must allow direct 911 dialing AND notify a designated person when 911 is dialed
Kari's Law — Genesys Cloud Voice Direct 911 dialing is automatic; notification requires setup with Genesys Voice support
Kari's Law — BYOC Customer must check with their carrier
Emergency number config Admin → Telephony → Sites → Number Plans → Emergency plan
BYOC warning Do not assign emergency number plan to BYOC trunk without verifying carrier support
HELD Protocol for Poly phones to deliver precise location to E911 — configured on trunk + base settings
Supported HELD phones Poly VVX · Poly CCX · Poly Edge E
Locations Admin → Telephony → Locations — stores physical addresses for E911 and Kari's Law

Telephony Connection Options — BYOC Cloud vs BYOC Premises

Section Description
Feature Area Telephony Infrastructure
Navigation Admin → Telephony → Trunks
Alt Navigation Menu → Digital and Telephony → Telephony → Trunks
Primary Function Connect Genesys Cloud to a third-party telecommunications carrier using SIP trunks

Genesys Cloud offers three telephony connection options. Understanding the differences between them — and between the two BYOC variants in particular — is a common exam topic.

Three Telephony Connection Options

Option Description Who Controls the Carrier?
Genesys Cloud Voice Genesys-provided telephony service over AWS infrastructure — fully managed by Genesys Genesys provides the carrier service
BYOC Cloud Customer brings their own carrier; SIP trunks terminate in Genesys Cloud's AWS-based Media Tier over the public internet Customer — uses their own carrier contract
BYOC Premises Customer brings their own carrier; SIP trunks terminate at a premises-based Edge hardware device Customer — uses their own carrier contract + their own on-premises Edge hardware

Study Notes

Topic Explanation
BYOC Bring Your Own Carrier — allows organizations to keep their existing carrier contract while using Genesys Cloud for contact center functionality
SIP Trunk A virtual phone line over IP that connects Genesys Cloud to the PSTN (public telephone network) via a carrier
BYOC Cloud Cloud-to-cloud — SIP trunks connect a third-party carrier directly to Genesys Cloud's Media Tier (AWS) over the public internet
BYOC Premises Hybrid — SIP trunks connect a third-party carrier to a Genesys Cloud Edge appliance installed on the customer's premises
Genesys Cloud Edge A physical hardware device installed on the customer's premises — required for BYOC Premises
Media Tier Genesys Cloud's AWS-based media processing infrastructure — where BYOC Cloud SIP trunks terminate
Trunk Type BYOC Cloud uses BYOC Carrier or BYOC PBX trunk types; BYOC Premises uses External SIP trunk type

BYOC Cloud vs BYOC Premises — Side-by-Side Comparison (Exam Critical)

Attribute BYOC Cloud BYOC Premises
Where SIP trunks terminate Genesys Cloud Media Tier (AWS, cloud) Genesys Cloud Edge (on-premises hardware)
Connectivity Over the public internet On-premises network + internet for cloud connectivity
Hardware required None — fully cloud-based Yes — Genesys Cloud Edge appliance
Trunk types used BYOC Carrier · BYOC PBX External SIP
Third-party device Can be a cloud-based carrier OR a premises-based carrier device / SBC Can connect to a premises-based carrier device (SBC/SIP gateway) or cloud-based carrier device
E911 Customer must verify E911 support with carrier Customer must verify E911 support with carrier
Kari's Law Customer must check with carrier for compliance Customer must check with carrier for compliance
BYOC Premises hardware deprecation N/A Genesys Hardware Solution end of support: December 1, 2026 (announced March 2025)

Trunk Types in Detail

Trunk Type Used With Description
BYOC Carrier BYOC Cloud SIP trunk to a third-party carrier (telephone company)
BYOC PBX BYOC Cloud SIP trunk to a third-party PBX (private branch exchange)
External SIP BYOC Premises SIP trunk from a premises-based Edge device to a third-party system
SIP Phone Trunk All Internal trunk type for SIP phones
WebRTC Phone Trunk All Internal trunk type for WebRTC phones

BYOC Cloud — How It Works

Third-party carrier (cloud or premises-based)
        ↓  (SIP trunk over public internet)
Genesys Cloud Media Tier (AWS)
        ↓
Genesys Cloud contact center features
        ↓
Agent desktop (WebRTC or managed/unmanaged phone)

Configuration: Admin → Telephony → Trunks → External Trunks → Add → BYOC Carrier or BYOC PBX

BYOC Premises — How It Works

Third-party carrier (cloud or on-premises SBC/gateway)
        ↓  (SIP trunk to Edge appliance)
Genesys Cloud Edge (on-premises hardware)
        ↓  (internet connection to Genesys Cloud)
Genesys Cloud contact center features
        ↓
Agent desktop (phone on same premises network or WebRTC)

Configuration: Admin → Telephony → Trunks → External Trunks → Add → External SIP

Genesys Cloud Edge (BYOC Premises Hardware)

The Edge is the on-premises appliance that bridges the customer's local SIP trunks to Genesys Cloud. It handles media processing and call control locally before relaying to the cloud.

Hardware Solution Notes
Genesys Hardware Solution Deprecated — End of Support: December 1, 2026
Customer-provided Edge (virtual or third-party hardware) Customers should plan migration to alternative Edge solutions before the EOS date

If you are on Genesys-provided BYOC Premises hardware, plan your migration strategy before December 1, 2026.

Emergency Services with BYOC

Unlike Genesys Cloud Voice (which includes built-in E911 via AWS), BYOC customers must work with their carrier for emergency services compliance:

Scenario E911 Approach
BYOC Cloud Check with your carrier — the carrier must support E911 for the numbers and locations in use
BYOC Premises Check with your carrier — same requirement; carrier must support E911
BYOC + Kari's Law (US) Check with your carrier — Genesys Cloud Voice handles this natively; BYOC requires carrier verification
BYOC Premises + Site configuration Configure emergency number plan in Admin → Telephony → Sites → Number Plans — do not assign an emergency number plan to a BYOC trunk unless the carrier has confirmed support

When to Use Each Option

Note: As of June 2025, Genesys deprecated Remote Survivability for BYOC Premises Edges, citing inability to reliably deliver IVR flows and AI features during internet outages.

Permissions

Permission Purpose
Telephony > Plugin > All Full access to telephony configuration
Telephony > SipTrunk > View/Add/Edit/Delete Trunk-specific permissions

Key Takeaways

Topic Summary
Three options Genesys Cloud Voice · BYOC Cloud · BYOC Premises
BYOC Cloud Carrier SIP trunks → Genesys AWS Media Tier · No on-premises hardware · Trunk types: BYOC Carrier / BYOC PBX
BYOC Premises Carrier SIP trunks → on-premises Edge appliance → Genesys Cloud · Requires Edge hardware · Trunk type: External SIP
Key distinction Where SIP trunks terminate — cloud (BYOC Cloud) vs on-premises Edge (BYOC Premises)
E911 Both BYOC options require carrier verification — not built-in like Genesys Cloud Voice
Premises hardware deprecation Genesys Hardware Solution for BYOC Premises: EOS December 1, 2026
Remote Survivability Deprecated as of June 2025 — no longer supported for BYOC Premises Edges