Setting up WebRTC phones for Genesys Cloud agents

See also

Sources

1. 1
   Active persistent connection management establishes the persistent connection before agents can accept calls and momentarily prevents an agent from going on queue, or calls from routing to them, until the connection is active.

   Configure a phone for active persistent connection · Opening section · Checked 2026-10-02
2. 2
   The active persistent connection article names Telephony > Plugin > All as the permission needed to enable modern WebRTC and active persistent connection management in Global Telephony Settings.disputed

   Configure a phone for active persistent connection · Prerequisites / permission line · Checked 2026-10-02
3. 3
   Enable modern WebRTC must be selected before Enable active persistent connection management can be selected.

   Configure a phone for active persistent connection · Prerequisites · Checked 2026-10-02
4. 4
   Organisations using SIP hardware phones, SIP softphones or remote phones must enable modern WebRTC to use active persistent connection even if they use no WebRTC phones.

   Configure a phone for active persistent connection · Prerequisites, note on non-WebRTC phones · Checked 2026-10-02
5. 5
   For BYOC Premises WebRTC problems Genesys tells admins to confirm WebRTC phone trunks are assigned to the Edge groups and that all Edges run the latest software version.

   Troubleshoot the Genesys Cloud WebRTC phone (admins) · Admin checklist, BYOC Premises items · Checked 2026-10-02
6. 6
   Organisations on BYOC Cloud or Genesys Cloud Voice must not select the PureCloud Voice - AWS site when creating phones, because it is a placeholder managed site that cannot be viewed or configured for phones.

   Can I use the PureCloud Voice – AWS site when creating phones? · FAQ answer · Checked 2026-10-02
7. 7
   WebRTC base settings are created under Menu > Digital and Telephony > Telephony > Phone Management on the Base Settings tab by clicking Add, naming the configuration and choosing Genesys Cloud WebRTC Phone in the Phone Make and Model list.

   Configure the Genesys Cloud WebRTC phone · Step 1: Create base settings · Checked 2026-10-02
8. 8
   The WebRTC phone requirements article tells users to run the latest Chrome or Firefox and to turn off the browser pop-up blocker so the separate WebRTC window that maintains phone connectivity can open.

   Requirements for WebRTC phones · Browser requirements section · Checked 2026-10-02
9. 9
   Before running the standalone WebRTC Diagnostics app the user must have voicemail configured and must disconnect any active persistent connection.

   Run the standalone Genesys Cloud WebRTC Diagnostics app · Prerequisites · Checked 2026-10-02
10. 10
    The standalone WebRTC Diagnostics app's phone test estimates call quality with MOS (1-5) and reports packet loss, round-trip time and jitter against targets of under 1 percent, under 300 ms and under 30 ms, and its network test checks DNS and reachability of Genesys media servers.

    Run the standalone Genesys Cloud WebRTC Diagnostics app · WebRTC Phone Test and Network Test sections · Checked 2026-10-02
11. 11
    The standalone WebRTC Diagnostics app is reached at a region-specific URL (for US East, https://apps.mypurecloud.com/webrtc-troubleshooter/) and requires logging in to Genesys Cloud.

    Run the standalone Genesys Cloud WebRTC Diagnostics app · How to run it; login requirement · Checked 2026-10-02
12. 12
    WebRTC base settings expose a DSCP (RTP) quality-of-service value from 00 to 3F in the Media section and a Signaling DSCP (SIP) value in the Network section.disputed

    Configure the Genesys Cloud WebRTC phone · Step 1, Media section DSCP (RTP); Network section Signaling DSCP (SIP) · Checked 2026-10-02
13. 13
    For BYOC Premises, a WebRTC or SIP phone trunk is attached to an Edge group under Menu > Digital and Telephony > Telephony > Edge Groups via Phone Trunks > Select Phone Trunk.

    Create a BYOC Premises Edge group with a phone trunk · Procedure, Phone Trunks section · Checked 2026-10-02
14. 14
    By default an Edge inherits the phone trunks selected for its Edge group, shown as Inherit from Edge Group on the Edge's Network Interfaces tab.

    Create a BYOC Premises Edge group with a phone trunk · Trunk inheritance paragraph · Checked 2026-10-02
15. 15
    Genesys lists Poly VVX 300-600 series, SoundPoint IP, SoundStation IP 4000/7000, RealPresence Trio 8500 and AudioCodes 420HD, 430HD and 440HD as end-of-life third-party managed phones that it cannot guarantee will support all new features.

    Genesys Cloud EOL position statement on third-party managed phones · EOL device models list · Checked 2026-10-02
16. 16
    Genesys recommends the Genesys Cloud WebRTC phone for organisations without strict physical-phone requirements, and newer supported models otherwise, in place of end-of-life third-party managed phones.

    Genesys Cloud EOL position statement on third-party managed phones · Recommended alternatives section · Checked 2026-10-02
17. 17
    Under Genesys Cloud Voice the external trunk is enabled by default, so configuring a WebRTC phone needs only base settings and a phone.

    Configure the Genesys Cloud WebRTC phone · Configure the Genesys Cloud WebRTC phone, prerequisites paragraph (Genesys Cloud Voice note) · Checked 2026-10-02
18. 18
    Genesys Cloud documents WebRTC headset integrations for Plantronics/Poly, Jabra, Sennheiser/EPOS, Yealink and Cyber Acoustics headsets.

    About WebRTC · Configuration scope / headset integration links · Checked 2026-10-02
19. 19
    Jabra call controls work in Chromium browsers (Chrome or Edge), the Genesys Cloud desktop app and the embedded clients, not in Firefox or Safari, and in a browser the headset is paired through a WebHID connection prompt.

    Configure a Jabra headset · Supported platforms and Chromium browser setup sections · Checked 2026-10-02
20. 20
    Jabra headset call controls with the Genesys Cloud WebRTC phone need the headset on USB, directly or through a Jabra USB Bluetooth adapter; Jabra headsets paired through the computer's internal Bluetooth adapter get no call control.

    Configure a Jabra headset · Call controls requirements section · Checked 2026-10-02
21. 21
    Configuring a managed phone consists of creating a base settings configuration and then creating the phone that uses it.

    About managed phone configuration · Opening paragraph · Checked 2026-10-02
22. 22
    Creating a managed Genesys, AudioCodes or Poly phone requires the phone name, base settings, site and the device MAC address entered as a single string, plus a line appearance on the Line Keys tab.

    Create a phone · Phone tab and Line Keys tab sections · Checked 2026-10-02
23. 23
    To use Media Helper an agent opens the region-specific Media Helper URL outside the VDI (for US East, https://apps.mypurecloud.com/mediahelper), logs in and allows microphone access, then logs in to Genesys Cloud inside the VDI with a WebRTC phone selected.

    Use WebRTC Media Helper in a Virtual Desktop Infrastructure (VDI) environment · Access the WebRTC Media Helper app; Choose your region table · Checked 2026-10-02
24. 24
    Closing the WebRTC Media Helper browser tab ends the Media Helper session.

    Use WebRTC Media Helper in a Virtual Desktop Infrastructure (VDI) environment · Media Helper tab note · Checked 2026-10-02
25. 25
    WebRTC Media Helper runs WebRTC media outside the VDI session, terminating audio on the client workstation while the Genesys Cloud interface runs inside the VDI, and can also be delivered by Browser Content Redirection where the VDI vendor supports it.

    WebRTC Media Helper overview · Overview and how it works sections · Checked 2026-10-02
26. 26
    WebRTC Media Helper does not work as a standalone WebRTC client and does not check for audio devices.

    WebRTC Media Helper overview · Limitations section · Checked 2026-10-02
27. 27
    The WebRTC Media Helper overview lists enabling the Require WebRTC Media Helper setting among the configuration requirements, together with a WebRTC phone created for and selected by each agent.

    WebRTC Media Helper overview · WebRTC Media Helper overview, requirements list · Checked 2026-10-02
28. 28
    Genesys directs administrators to install and configure Microsoft multimedia redirection by following Microsoft's documentation before the Genesys Cloud side is configured.

    Configure Microsoft's Multimedia Redirection to use the Genesys Cloud WebRTC phone · Section 'Microsoft multimedia redirection' · Checked 2026-10-02
29. 29
    To use Microsoft Multimedia Redirection with the Genesys Cloud WebRTC phone on Azure Virtual Desktop or Windows 365 Cloud PC, Require WebRTC Media Helper must be set to Disabled in the WebRTC base settings or phone settings.

    Configure Microsoft's Multimedia Redirection to use the Genesys Cloud WebRTC phone · Section 'Genesys Cloud WebRTC Media Helper' · Checked 2026-10-02
30. 30
    With modern WebRTC and persistent connection, the client keeps microphone access between calls but is put on hold and sends no RTP media, exchanging only keep-alive messages until the next call.

    Does Genesys Cloud continue collecting audio in between WebRTC calls when using the Modern WebRTC feature and persistent connection? · FAQ answer, 'With Modern WebRTC enabled' paragraph · Checked 2026-10-02
31. 31
    Disabling modern WebRTC automatically disables active persistent connection management.

    Enable modern WebRTC · Limitations · Checked 2026-10-02
32. 32
    The Enable modern WebRTC article names Telephony > Settings > All as the permission needed to change Global Telephony Settings.disputed

    Enable modern WebRTC · Prerequisites / permission line · Checked 2026-10-02
33. 33
    Modern WebRTC is enabled under Menu > Digital and Telephony > Telephony > Global Telephony Settings by selecting Enable modern WebRTC under Voice Calls and saving.

    Enable modern WebRTC · How to enable modern WebRTC, steps 1-3 · Checked 2026-10-02
34. 34
    After enabling modern WebRTC, administrators must wait 24 hours for the configuration to roll out to all WebRTC phones before enabling active persistent connection or the Idle Media attempt-recovery feature.

    Enable modern WebRTC · Limitations / note on 24-hour rollout · Checked 2026-10-02
35. 35
    MOS values for WebRTC calls, on a 1 to 5 scale, are viewable in the Genesys Cloud Interactions view.

    Does Genesys Cloud support QoS for WebRTC phones? · FAQ answer, MOS paragraph · Checked 2026-10-02
36. 36
    A Genesys FAQ states that Genesys Cloud does not natively support QoS for WebRTC phones and offers Mean Opinion Score as the quality measure instead.disputed

    Does Genesys Cloud support QoS for WebRTC phones? · FAQ answer, first paragraph · Checked 2026-10-02
37. 37
    Only one WebRTC phone can be created per user in Genesys Cloud.

    Configure the Genesys Cloud WebRTC phone · Configure the Genesys Cloud WebRTC phone, prerequisites paragraph · Checked 2026-10-02
38. 38
    In WebRTC base settings the Preferred Codec note states that WebRTC only works with the Opus codec.

    Configure the Genesys Cloud WebRTC phone · Step 1, Media section, Preferred Codec · Checked 2026-10-02
39. 39
    Configuring Genesys Cloud WebRTC phones requires the Telephony > Plugin > All permission.

    Configure the Genesys Cloud WebRTC phone · Configure the Genesys Cloud WebRTC phone, prerequisites paragraph · Checked 2026-10-02
40. 40
    An agent can end an active persistent connection from the Calls panel via Phone Details, the phone selector drop-down and Disconnect Persistent Connection, and Genesys Cloud re-establishes it on the agent's next call.

    Terminate a persistent connection on a Genesys Cloud WebRTC phone · Procedure steps 1-4 and closing note · Checked 2026-10-02
41. 41
    Over an established persistent connection the UI alerts the agent immediately, or answers the call immediately if the user has Auto Answer configured.

    Use the persistent connection feature with a Genesys Cloud WebRTC phone · Auto answer paragraph · Checked 2026-10-02
42. 42
    While a persistent connection is being re-established, Genesys Cloud may assign the agent non-voice interactions such as chat, email and messages from their queues.

    Use the persistent connection feature with a Genesys Cloud WebRTC phone · Non-voice interactions paragraph · Checked 2026-10-02
43. 43
    Genesys recommends enabling persistent connection outside business hours, and agents must log out and back in after it is enabled.

    Configure the Genesys Cloud WebRTC phone · Step 1, General section, Persistent Connection note · Checked 2026-10-02
44. 44
    A failed persistent connection puts the agent into a Not Responding state across all channels, including agents who handle only digital interactions.

    Use the persistent connection feature with a Genesys Cloud WebRTC phone · Caveat paragraph · Checked 2026-10-02
45. 45
    With persistent connection, the WebRTC phone's connection stays open after a call ends so Genesys Cloud can pass the next call to the agent over the existing connection, and a green indicator appears next to the phone icon.

    Use the persistent connection feature with a Genesys Cloud WebRTC phone · How persistent connection works · Checked 2026-10-02
46. 46
    Genesys Cloud initiates the persistent connection when a user with the routing:conversation:acceptOffQueue permission becomes Available off-queue, or otherwise when the user goes On Queue.

    Use the persistent connection feature with a Genesys Cloud WebRTC phone · Activation conditions paragraph · Checked 2026-10-02
47. 47
    The WebRTC persistent connection timeout defaults to 600 seconds (10 minutes) and can be set up to 604,800 seconds (7 days).

    Configure the Genesys Cloud WebRTC phone · Step 1, General section, Persistent Connection setting · Checked 2026-10-02
48. 48
    If no call arrives before the persistent connection timeout expires, the connection terminates and is re-established automatically, and the agent can still receive calls.

    Use the persistent connection feature with a Genesys Cloud WebRTC phone · Timeout paragraph · Checked 2026-10-02
49. 49
    A WebRTC phone is created on the Phones tab with Add Phone by entering a Phone Name, selecting the base settings, choosing a Site, selecting the user in the Person list, and clicking Save Phone.

    Configure the Genesys Cloud WebRTC phone · Step 2: Create the phone · Checked 2026-10-02
50. 50
    Changes to a WebRTC phone's settings take effect after the user logs out and logs back in.

    Configure the Genesys Cloud WebRTC phone · Step 2: Create the phone, closing note · Checked 2026-10-02
51. 51
    A phone inherits its base settings but can be customised per phone without changing the base settings configuration.

    Configure the Genesys Cloud WebRTC phone · Step 2: Create the phone · Checked 2026-10-02
52. 52
    UDP/19302 (STUN) to Google is listed as optional for WebRTC phones.

    Ports and services for WebRTC phones under Genesys Cloud Voice · WebRTC Signaling table, udp/19302 row · Checked 2026-10-02
53. 53
    WebRTC phone media (SRTP/TURN) uses UDP ports 16384-65535 to the Genesys Cloud Media Tier.

    Ports and services for WebRTC phones under Genesys Cloud Voice · WebRTC Media table, udp/16384-65535 row · Checked 2026-10-02
54. 54
    Under BYOC Premises, Edges connect to Genesys Cloud services for WebRTC phones over TCP/5061.

    Ports and services for WebRTC phones under BYOC Premises · WebRTC Cloud table, tcp/5061 row · Checked 2026-10-02
55. 55
    Under BYOC Premises the STUN ports TCP/3478 and UDP/3478 must be opened for both the client and the Edges.

    Ports and services for WebRTC phones under BYOC Premises · WebRTC Signaling table, tcp/3478 and udp/3478 rows · Checked 2026-10-02
56. 56
    WebRTC phone signaling uses TCP/443 (HTTPS) to Genesys Cloud and Amazon AWS.

    Ports and services for WebRTC phones under Genesys Cloud Voice · WebRTC Signaling table, tcp/443 row · Checked 2026-10-02
57. 57
    UDP/3478 (STUN) must be open for WebRTC phones because it carries the srflx and relay candidates, and Genesys warns that calls will have a high rate of failure if it is closed.

    Ports and services for WebRTC phones under Genesys Cloud Voice · WebRTC Signaling table, udp/3478 row · Checked 2026-10-02
58. 58
    Enabling Require WebRTC Media Helper prevents failback to streaming media through the VDI session, and subsequent calls fail until the Media Helper is re-authenticated.

    Configure the Genesys Cloud WebRTC phone · Step 1, Media section, Require WebRTC Media Helper · Checked 2026-10-02
59. 59
    The Require WebRTC Media Helper setting in WebRTC base settings is disabled by default.

    Configure the Genesys Cloud WebRTC phone · Step 1, Media section, Require WebRTC Media Helper · Checked 2026-10-02
60. 60
    In the Calls panel Settings an agent can select the microphone and speaker, adjust device volumes, and use Run diagnostics to check the microphone, connectivity and throughput.

    Select and configure the Genesys Cloud WebRTC phone · Configure WebRTC phone settings section · Checked 2026-10-02
61. 61
    An agent selects the WebRTC phone by opening Calls, clicking Phone Details, typing the phone name in Search for phone and choosing it from the suggestions; Phone Details shows red with a line through it when no phone is assigned.

    Select and configure the Genesys Cloud WebRTC phone · Select the WebRTC phone, steps 1-4 · Checked 2026-10-02
62. 62
    The SIP phone trunk procedure most likely applies to managed phones registering to BYOC Premises Edges, because its listen ports match the Edge signaling ports in the BYOC Premises VoIP port list.inferred

    Create a SIP phone trunk · Procedure step 5 (listen port), compared with Signaling table of the BYOC Premises VoIP ports page · Checked 2026-10-02
63. 63
    A SIP phone trunk is created on the Phone Trunks tab with Type SIP, Trunk State In-Service, a protocol and listen port (8060 for UDP or TCP, 8061 for TLS), and SIP access control by source address listing phone subnets in CIDR notation.

    Create a SIP phone trunk · Procedure steps 1-10 · Checked 2026-10-02
64. 64
    The WebRTC phone Test Settings button checks only the microphone; broader testing launches the Genesys Cloud WebRTC Diagnostics app.

    Test your WebRTC phone settings · Diagnostic process section · Checked 2026-10-02
65. 65
    A WebRTC phone trunk is created under Menu > Digital and Telephony > Telephony > Trunks on the Phone Trunks tab with Create New, choosing Type Cloud WebRTC Phone Connections for BYOC Cloud or Premises WebRTC Phone Connections for BYOC Premises.

    Create a WebRTC phone trunk · Procedure steps 1-5 · Checked 2026-10-02
66. 66
    The TURN Behavior media setting reduces the number of open outbound firewall ports by relaying through automatically scaling TURN service instances and requires a free subscription add-on.

    Configure the Genesys Cloud WebRTC phone · Step 1, Media section, TURN Behavior · Checked 2026-10-02
67. 67
    Under Genesys Cloud Voice, managed VoIP phones signal to the Genesys Cloud Media Tier on TCP/8061 (SIPS), while unmanaged phones use TCP/5061, and media uses UDP 16384-65535 to the Media Tier.

    Ports and services for VoIP phones under Genesys Cloud Voice · Signaling and Media tables · Checked 2026-10-02
68. 68
    Under BYOC Premises, managed VoIP phones send configuration, signaling (UDP/TCP 8060 SIP, TCP/8061 SIPS) and media (UDP 16384-65535) to the Genesys Cloud Edge devices rather than the Media Tier.

    Ports and services for VoIP phones under BYOC Premises · Configuration Provisioning, Signaling and Media tables · Checked 2026-10-02
69. 69
    Managed VoIP phones fetch configuration over TCP/80 and TCP/443 from Genesys Cloud Global Phone Provisioning (AWS), and unconfigured Polycom phones use TCP/443 to Polycom zero-touch provisioning.

    Ports and services for VoIP phones under Genesys Cloud Voice · Zero-Touch Provisioning and Configuration Provisioning tables · Checked 2026-10-02

Documents

Cite this page

WarmTransfer. (2026, October 2). Setting up WebRTC phones for Genesys Cloud agents. WarmTransfer. https://warmtransfer.net/guides/genesys-cloud-webrtc-phone-setup

@misc{warmtransfer-genesys-cloud-webrtc-phone-setup,
  title  = {Setting up WebRTC phones for Genesys Cloud agents},
  author = {{WarmTransfer}},
  year   = {2026},
  url    = {https://warmtransfer.net/guides/genesys-cloud-webrtc-phone-setup},
  note   = {Verified 2026-10-02}
}