WebRTC in modern contact centers

See also

Sources

1. 1
   AWS's WebRTC best-practices text tells customers to allow-list incoming UDP traffic on port 3478, while the networking page lists 3478 only as an outbound rule and says no inbound client port is needed.disputed

   How Contact Control Panel (CCP) uses WebRTC · Best practices, second bullet · Checked 2026-09-24
2. 2
   AWS strongly recommends that CCP media between the agent workstation and AWS go directly, without traversing VPNs or network accelerator hops.

   How Contact Control Panel (CCP) uses WebRTC · Best practices, first bullet · Checked 2026-09-24
3. 3
   AWS states that the CCP softphone uses persistent outbound send/receive connections from the browser, so no client-side port has to be opened for inbound traffic.

   Set up your network to use the Connect Customer Contact Control Panel (CCP) · Introductory paragraph of 'Set up your network to use the ... CCP' · Checked 2026-09-24
4. 4
   The Amazon Connect CCP sends its SDP offer, including host, STUN-derived and TURN relay candidates, to Amazon Connect over a WebSocket signaling channel, and receives the SDP answer the same way.

   How Contact Control Panel (CCP) uses WebRTC · How STUN, TURN and ICE work together, steps 1-3 · Checked 2026-09-24
5. 5
   With a stateless firewall, Amazon Connect also requires allowing inbound UDP to the browser's ephemeral port range from AMAZON_CONNECT ranges (49152-65535 on Windows, 32768-61000 on Linux).

   Set up your network to use the Connect Customer Contact Control Panel (CCP) · Stateless firewalls section table · Checked 2026-09-24
6. 6
   Amazon Connect CCP softphone media requires outbound UDP 3478 (send/receive) to the regional TurnNlb-*.elb.<region>.amazonaws.com TURN endpoints.

   Set up your network to use the Connect Customer Contact Control Panel (CCP) · Option 1 (recommended) allowlist table, TurnNlb row; NLB endpoints table · Checked 2026-09-24
7. 7
   For Amazon WorkSpaces, Citrix, Omnissa, Azure Virtual Desktop or Windows 365, AWS says a custom CCP built with its open-source libraries can offload audio processing to the agent's local device and redirect audio to Amazon Connect.

   Use Connect Customer in a VDI environment · Cloud desktops · Checked 2026-09-24
8. 8
   For VDI, AWS recommends a split CCP: a media-less CCP inside the virtual desktop for data and call control, and a standard CCP carrying media on the local PC.

   Use Connect Customer in a VDI environment · Use a split CCP model · Checked 2026-09-24
9. 9
   WebRTC implementations must periodically verify continued consent to send using the RFC 7675 consent freshness mechanism (ICE Binding Requests), not plain STUN keepalives.

   RFC 8827: WebRTC Security Architecture · Section 4.4 (Communications and Consent Freshness); Section 6.3 (Communications Consent) · Checked 2026-09-24
10. 10
    WebRTC implementations should attempt to set DSCP QoS markings per RFC 8837, and may stop marking if they detect blocking or priority inversion.

    RFC 8835: Transports for WebRTC · Section 4.2 (Usage of Quality of Service -- DSCP and Multiplexing) · Checked 2026-09-24
11. 11
    All WebRTC implementations must support DTLS 1.2 with the TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite.

    RFC 8827: WebRTC Security Architecture · Section 6.5 (Communications Security) · Checked 2026-09-24
12. 12
    WebRTC endpoints must implement full ICE (RFC 8445), not ICE-Lite.

    RFC 8835: Transports for WebRTC · Section 3.4 (Middlebox-Related Functions) · Checked 2026-09-24
13. 13
    Genesys's WebRTC overview page says the WebRTC phone works in Chrome, Firefox or Microsoft Edge (Chromium), in the desktop or web app.disputed

    WebRTC overview · Overview body text · Checked 2026-09-24
14. 14
    Genesys's WebRTC phone requirements page says to use the latest version of Chrome or Firefox.disputed

    Requirements for WebRTC phones · Browsers requirement · Checked 2026-09-24
15. 15
    Genesys Cloud can force WebRTC ICE to TURN relay candidates; Genesys says this shortens negotiation but reduces resiliency, and requires allow-listing four published CIDR blocks.

    Force the Relay/TURN Behavior · Force the Relay/TURN Behavior: overview and CIDR allowlist section · Checked 2026-09-24
16. 16
    Under Genesys Cloud Voice, WebRTC phone audio (SRTP/TURN) uses UDP 16384-65535 to the Genesys Cloud Media Tier.

    Ports and services for WebRTC phones under Genesys Cloud Voice · Ports table, WebRTC Media row · Checked 2026-09-24
17. 17
    Genesys Cloud does not support customer-run private STUN/TURN servers or private proxy servers for its WebRTC connections.

    Can we implement our own STUN/TURN server to proxy connections to Genesys Cloud? · FAQ answer · Checked 2026-09-24
18. 18
    The Genesys Cloud WebRTC phone needs the browser pop-up blocker off, and for Embeddable Framework, Salesforce, Zendesk and Microsoft Teams integrations it runs in a separate browser window that holds the WebRTC connection.

    Requirements for WebRTC phones · Pop-up blocker and separate WebRTC window requirements · Checked 2026-09-24
19. 19
    Under Genesys Cloud Voice, the WebRTC phone uses UDP 3478 STUN for server-reflexive and relay candidates, and Genesys warns calls will have a high failure rate if it is closed.

    Ports and services for WebRTC phones under Genesys Cloud Voice · Ports table, WebRTC Signaling UDP/3478 row · Checked 2026-09-24
20. 20
    WebRTC implementations must refuse all device permission grants to plain HTTP origins, so browser agent clients must be served over HTTPS.

    RFC 8827: WebRTC Security Architecture · Section 6.2 (Device Permissions Model) · Checked 2026-09-24
21. 21
    JSEP does not specify a signaling model or state machine; the application decides which offers and answers to pass to the browser and when.

    RFC 9429: JavaScript Session Establishment Protocol (JSEP) · Section 3.1 (Signaling Model); Section 1.1 (General Design of JSEP) · Checked 2026-09-24
22. 22
    RFC 9429 (JSEP, April 2024) obsoletes RFC 8829, mainly to fix inconsistencies in BUNDLE handling found after RFC 8829 was published.

    RFC 9429: JavaScript Session Establishment Protocol (JSEP) · Abstract; header 'Obsoletes: 8829' · Checked 2026-09-24
23. 23
    JSEP session descriptions use SDP syntax, carried in a SessionDescription object that can be built from and serialised to SDP.

    RFC 9429: JavaScript Session Establishment Protocol (JSEP) · Section 3.3 (Session Description Format) · Checked 2026-09-24
24. 24
    JSEP supports optional ICE candidate trickling: the application can send the offer immediately and send candidates as they are gathered.

    RFC 9429: JavaScript Session Establishment Protocol (JSEP) · Section 3.5.2 (ICE Candidate Trickling) · Checked 2026-09-24
25. 25
    WebRTC implementations must obtain explicit user consent before giving an application access to the camera or microphone.

    RFC 8827: WebRTC Security Architecture · Section 6.2 (Device Permissions Model) · Checked 2026-09-24
26. 26
    WebRTC endpoints must implement Opus (RFC 7587 payload format) and G.711 PCMA and PCMU.

    RFC 7874: WebRTC Audio Codec and Processing Requirements · Section 3 (Codec Requirements) · Checked 2026-09-24
27. 27
    RFC 8825 defines a WebRTC non-browser endpoint as one that implements the WebRTC protocols but not the JavaScript API, distinct from a WebRTC browser that implements both.

    RFC 8825: Overview: Real-Time Protocols for Browser-Based Applications · Terminology definitions 'WebRTC Browser', 'WebRTC Non-Browser', 'WebRTC Endpoint' · Checked 2026-09-24
28. 28
    RFC 3389 comfort noise is required for G.711 but not recommended with Opus, which has its own comfort-noise mechanism.

    RFC 7874: WebRTC Audio Codec and Processing Requirements · Section 3 (Codec Requirements) · Checked 2026-09-24
29. 29
    For endpoints handling audio above 8 kHz, RFC 7874 recommends offering Opus before PCMA/PCMU.

    RFC 7874: WebRTC Audio Codec and Processing Requirements · Section 3 (Codec Requirements) · Checked 2026-09-24
30. 30
    RFC 8827 requires the WebRTC API to let the application restrict ICE to TURN relay candidates so that the peer does not learn the user's IP addresses.

    RFC 8827: WebRTC Security Architecture · Section 6.4 (IP Location Privacy) · Checked 2026-09-24
31. 31
    RFC 7874 allows DTMF event durations of 40 to 8000 ms (100 ms recommended) with a minimum 30 ms gap between events (70 ms recommended).

    RFC 7874: WebRTC Audio Codec and Processing Requirements · Section 3 (Codec Requirements), DTMF paragraph · Checked 2026-09-24
32. 32
    RFC 8825 is an applicability statement: it does not itself specify a protocol but names the specifications a WebRTC implementation must follow.

    RFC 8825: Overview: Real-Time Protocols for Browser-Based Applications · Abstract · Checked 2026-09-24
33. 33
    RFC 8825's list of required specifications names RFC 8829 for JSEP, which RFC 9429 has since obsoleted, so a reader following RFC 8825 literally lands on a superseded JSEP.inferred

    RFC 8825: Overview: Real-Time Protocols for Browser-Based Applications · Normative References (RFC 8829 entry), read with RFC 9429 header · Checked 2026-09-24
34. 34
    WebRTC implementations are required to support multiplexing RTP and RTCP on a single transport flow (RFC 5761).

    RFC 8834: Media Transport and Use of RTP in WebRTC · Section 4.5 (RTP and RTCP Multiplexing) · Checked 2026-09-24
35. 35
    WebRTC endpoints must not send using the basic RTP/AVP profile; they must use the RTP/SAVPF profile.

    RFC 8834: Media Transport and Use of RTP in WebRTC · Section 4.2 (Choice of the RTP Profile) · Checked 2026-09-24
36. 36
    WebRTC implementations must not offer SDP security descriptions (SDES, RFC 4568) nor select SDES if it is offered; keying is DTLS-SRTP.

    RFC 8827: WebRTC Security Architecture · Section 6.5 (Communications Security) · Checked 2026-09-24
37. 37
    The choice of client-server and inter-server signaling protocols, and the translation between them, is outside the scope of the WebRTC protocol suite.

    RFC 8825: Overview: Real-Time Protocols for Browser-Based Applications · sentence beginning 'The choice of protocols for client-server and inter-server signaling' (section number not captured) · Checked 2026-09-24
38. 38
    RFC 7118 defines SIP over WebSocket using the WebSocket subprotocol name 'sip', letting browser clients act as SIP user agents.

    RFC 7118: The WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP) · WebSocket subprotocol negotiation text and IANA Considerations (subprotocol registration) · Checked 2026-09-24
39. 39
    RFC 7118 defines Via transport values WS for plain WebSocket and WSS for secure WebSocket, and recommends that SIP over WebSocket be protected with WSS (TLS).

    RFC 7118: The WebSocket Protocol as a Transport for the Session Initiation Protocol (SIP) · Via transport parameter definition; Security Considerations (RECOMMENDED secure WebSocket) · Checked 2026-09-24
40. 40
    All WebRTC media channels must be secured with SRTP and SRTCP; plaintext RTP/RTCP must not be sent and null cipher suites must not be negotiated.

    RFC 8827: WebRTC Security Architecture · Section 6.5 (Communications Security) · Checked 2026-09-24
41. 41
    WebRTC endpoints must implement audio/telephone-event (RFC 4733) and be able to generate and consume DTMF events 0-9, *, # and A-D.

    RFC 7874: WebRTC Audio Codec and Processing Requirements · Section 3 (Codec Requirements) · Checked 2026-09-24
42. 42
    WebRTC endpoints must support TURN, including TURN over TCP and TURN over TLS over TCP, so media can traverse firewalls that block UDP.

    RFC 8835: Transports for WebRTC · Section 3.4 (Middlebox-Related Functions) · Checked 2026-09-24
43. 43
    Among the vendors examined, browser WebRTC audio inside VDI is either unsupported (Webex Contact Center) or handled by moving media to the local device (Amazon Connect split CCP or audio offload), pointing to hairpinned VDI audio as a shared design constraint.inferred

    Use Connect Customer in a VDI environment · Use a split CCP model; Cloud desktops; read with Webex CC system requirements VDI statement · Checked 2026-09-24
44. 44
    Because standard WebRTC leaves signaling to the application, each CCaaS browser agent client uses its vendor's own signaling to its cloud (AWS documents a WebSocket channel); none of the fetched vendor pages say their browser client uses SIP over WebSocket.inferred

    How Contact Control Panel (CCP) uses WebRTC · How STUN, TURN and ICE work together, step 2; read with RFC 8825 signaling scope statement · Checked 2026-09-24
45. 45
    The W3C WebRTC API exposes this as RTCIceTransportPolicy with values 'relay' (TURN relay candidates only) and 'all'.

    WebRTC: Real-Time Communication in Browsers · RTCIceTransportPolicy enum definition · Checked 2026-09-24
46. 46
    The W3C RTCDTMFSender insertDTMF method accepts tone durations of 40 to 6000 ms and an inter-tone gap of at least 30 ms.

    WebRTC: Real-Time Communication in Browsers · RTCDTMFSender interface, insertDTMF method · Checked 2026-09-24
47. 47
    WebRTC: Real-Time Communication in Browsers (the RTCPeerConnection API) is a W3C Recommendation; the version at /TR/webrtc/ on 2026-09-24 is dated 13 March 2025.

    WebRTC: Real-Time Communication in Browsers · Status of This Document / document header · Checked 2026-09-24
48. 48
    Cisco states that Webex Contact Center WebRTC requires a minimum network bandwidth of 100 kbps per call.

    Set up voice channels for Webex Contact Center · WebRTC section, network requirement sentence · Checked 2026-09-24
49. 49
    Webex Contact Center WebRTC supports Google Chrome, Microsoft Edge and Mozilla Firefox.

    System requirements for Webex Contact Center Desktop · Minimum system requirements section · Checked 2026-09-24
50. 50
    Cisco advises running the Advanced CScan tool from every network where WebRTC agents will work, including call centres and remote or work-from-home agents.

    Set up voice channels for Webex Contact Center · WebRTC section, CScan paragraph · Checked 2026-09-24
51. 51
    A Webex Contact Center agent receives calls in the browser by choosing the Desktop telephony option under Station Credentials when signing in to Agent Desktop.

    Set up voice settings for Webex Contact Center · WebRTC section · Checked 2026-09-24
52. 52
    To give Webex Contact Center agents WebRTC, an administrator enables WebRTC under Tenant Settings > Voice in Control Hub and selects the Desktop option in the agent's desktop profile Voice Channel options.

    Set up voice channels for Webex Contact Center · 'Set up voice channels for Webex Contact Center', WebRTC / Enable WebRTC section · Checked 2026-09-24
53. 53
    As of 2026-09-24, Cisco states WebRTC Desktop support is unavailable for Webex Contact Center customers hosted in its Japan data centre.

    Set up voice channels for Webex Contact Center · WebRTC section, limitation note · Checked 2026-09-24
54. 54
    Webex Contact Center WebRTC delivers voice media to the agent's browser over the internet without needing a local gateway.

    Set up voice channels for Webex Contact Center · WebRTC section · Checked 2026-09-24
55. 55
    Webex Contact Center Agent Desktop is supported in VDI, but WebRTC calling is not supported in VDI.

    System requirements for Webex Contact Center Desktop · Minimum system requirements; Agent Desktop and Supervisor Desktop sections · Checked 2026-09-24
56. 56
    In Webex Contact Center, WebRTC browser calling is enabled by default for a new tenant.

    Set up voice settings for Webex Contact Center · WebRTC section of 'Set up voice settings for Webex Contact Center' · Checked 2026-09-24
57. 57
    Cisco TAC documents troubleshooting Webex Contact Center WebRTC calls in Chrome through chrome://webrtc-internals, which shows RTCPeerConnection traces and getStats-based live statistics and can export a dump.

    Exploring WebRTC Data and Statistics in WxCC Agent Desktop · Sections on accessing WebRTC statistics and creating a WebRTC-Internals dump · Checked 2026-09-24

Documents

Cite this page

WarmTransfer. (2026, September 24). WebRTC in modern contact centers. WarmTransfer. https://warmtransfer.net/knowledge/webrtc-deployment

@misc{warmtransfer-webrtc-deployment,
  title  = {WebRTC in modern contact centers},
  author = {{WarmTransfer}},
  year   = {2026},
  url    = {https://warmtransfer.net/knowledge/webrtc-deployment},
  note   = {Verified 2026-09-24}
}