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 Trunks — configure SIP connectivity; TLS transport is selected per trunk Edges & Edge Groups — BYOC Premises media appliances that rely on CA trust Sites — telephony routing configuration 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 Call Routing & Message Routing — DID numbers are associated with inbound call routes Architect Overview — call flows are the assignment target for main inbound numbers Extensions — separate from DIDs; extensions are internal-only dialing numbers Architectural Build Order — DID numbers are configured in Phase 2 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: __ See Also Sites — Edges are assigned within site-based telephony design Trunks — External SIP trunks attach to Edges and are shared through Edge Groups Certificate Authorities — required for TLS trust between Edge and remote endpoints Topology — visualize Edge status and phone-to-edge assignments Architectural Build Order — Edges are built in Phase 2 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: Telephony > Plugin > All Telephony > Extensions > Add, Edit, View, and Delete 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: Search by extension number View which user each extension is assigned to Open the user's profile directly from search results Customize visible columns and row density 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: _Ext_Pool__ See Also DID & Toll-Free Numbers — external numbers; different from internal extensions User Profile Management — Contact Information is where extensions are assigned to users Divisions & Access Control — divisions are required when creating extension pools Architectural Build Order — extensions are assigned during Phase 3 (People) 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: Number normalization Number plan classification Outbound route selection Trunk settings In-service Edges (BYOC Premises) Destination site 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: __ See Also Trunks — create trunks before configuring outbound routes Locations & Floor Plans — locations must exist before creating sites Edges & Edge Groups — BYOC Premises sites use Edge assignments WebRTC Phone Management — Media Regions and Relay/TURN Behavior are configured here Architectural Build Order — Sites are built in Phase 2 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 Sites — configure telephony routing and dial plans Trunks — configure carrier/PBX SIP connectivity Edges & Edge Groups — manage BYOC Premises media appliances WebRTC Phone Management — manage softphone endpoints 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: __ See Also Sites — outbound routes are configured here and reference trunks Certificate Authorities — required for TLS trunk trust (BYOC Premises) Edges & Edge Groups — BYOC Premises trunks attach to Edges Architectural Build Order — trunks are built in Phase 2 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: Choose microphone and speaker device Adjust volume Run audio diagnostics 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: _WebRTC_ See Also Sites — Media Regions and Relay/TURN Behavior are configured here Topology — confirm phone-to-edge assignments and site relationships Architectural Build Order — phones are assigned in Phase 3 (People) 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) Contact Genesys Cloud Voice support 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: Full physical street address Used by E911 routing (Genesys Cloud Voice) Used by Kari's Law notification configuration Assigned to sites, phones, or users as appropriate 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 Scenario Recommended Option Fastest, simplest deployment with no existing carrier contract Genesys Cloud Voice Organization has an existing carrier contract they want to keep BYOC Cloud Organization needs on-premises media processing or local survivability (within a site) BYOC Premises Fully cloud-native strategy with own carrier BYOC Cloud Hybrid deployment needing both cloud and on-premises BYOC Premises (possibly in combination with other options) 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