Set up voice channels for Webex Contact Center

Publisher

Cisco Systems, Inc. (Webex Help Center)

URL

https://help.webex.com/en-us/article/nhy3codb/Set-Up-Voice-Channel-for-Webex-Contact-Center

Published

2026-06-25

Updated

unknown

Accessed

2026-09-04

HTTP status

200

License

Cisco proprietary documentation, all rights reserved; no-redistribution; short excerpts and locators only

Source notes citing this source

• Because Webex Contact Center accepts only G.711 and does no transcoding itself, the per-call bandwidth on the leg into its media edge should be the G.711 figure: about 87.2 kbps per direction on Ethernet at 20 ms by Cisco's generic calculation. Codec choice cannot reduce it on that leg. inferred in context
• Webex Contact Center supports only G.711 u-law and G.711 A-law on its voice media path. in context
• Webex Contact Center requires any codec other than G.711 (such as G.729, iLBC or Opus) to be transcoded before it reaches Webex Contact Center media edge locations. in context
• Webex Contact Center states that WebRTC needs a minimum of 100 kbps of network bandwidth per call. in context
• Webex Contact Center's SIP interoperability guidelines list SIP Delayed Offer as Supported. in context
• Webex Contact Center's SIP interoperability guidelines list SIP Early Media as Not Supported, marked as a future roadmap item. in context
• Webex Contact Center's SIP interoperability guidelines list SIP Early Offer as Not Supported, marked as a future roadmap item. in context
• Because Webex Contact Center does not support SIP early media, a deployment should not expect in-band ringback or announcements from Webex Contact Center before answer; any ringback a caller hears before answer would have to come from local generation upstream (gateway, SBC or carrier). inferred in context
• Webex Contact Center documents five ways to deliver a call to an agent: PSTN interconnectivity with a DID or mobile number, Webex Calling registered endpoints such as MPP devices and the Webex App, WebRTC in the agent desktop, premises PBX or premises PSTN connectivity, and Microsoft Teams agents reached over Local Gateway direct SIP routing. in context
• The Cisco Webex Calling Plan option for Webex Contact Center is documented as currently available for Webex Contact Center trials and in regions where Cisco supports the service or inbound toll-free numbers. in context
• Webex Calling Cloud Connected PSTN provides PSTN connectivity in over 65 countries through Certified Calling Providers, and requires a relationship with an authorised Cloud Connected PSTN partner for activation and billing. in context
• Webex Contact Center PSTN purchased from Cisco as a bundle is offered to customers in the contiguous United States and Canada and does not support international dialing. in context
• Webex Contact Center requires G.711 uLaw or G.711 aLaw on the PSTN leg. in context
• Webex Contact Center requires RFC 2833 DTMF on the PSTN leg. in context
• Webex Calling supports +E.164 number formats by default when Cloud Connected PSTN or Local Gateway is used, and Control Hub stores PSTN numbers in +E.164 format. in context
• Webex Contact Center does not support SIP early media on the PSTN leg. in context
• Webex Contact Center does not support SIP Early Offer on the PSTN leg and does support SIP Delayed Offer. in context
• Webex Calling Local Gateway lets a customer keep its existing on-premises PSTN carrier and is described as providing access in over 140 markets globally, with the customer supplying and configuring the session border controller. in context
• Webex Contact Center expects a packetization timer of 20 ms in the SIP SDP on the PSTN leg. in context
• A Webex Contact Center tenant must be connected to the PSTN and must have at least one method of delivering calls to agents. in context
• Any codec other than G.711, including G.729, iLBC and Opus, must be transcoded before the media reaches Webex Contact Center. in context
• Cisco advises limiting the number of PSTN carriers involved in routing a call to Webex Contact Center to no more than two, and treats chained forwarding across multiple sequential carriers as unsupported because of the latency it introduces. in context
• Webex Contact Center does not support SIP Early Media on PSTN ingress. in context
• A Menu or Collect Digits activity that always takes its no-input path on a BYO PSTN entry point, even though callers press keys, matches carrier digits not arriving as RFC2833 telephone-events, and the SDP and RTP should be checked first. inferred in context
• Webex Contact Center requires any codec other than G.711, such as G.729, iLBC or Opus, to be transcoded before the call reaches Webex Contact Center media edge locations. in context
• Because RFC2833 is the only DTMF method Webex Contact Center lists, a BYO PSTN path that delivers digits only as SIP INFO, KPML, unsolicited NOTIFY or in-band audio needs its gateway or SBC to convert them to RFC2833 before Webex Contact Center. inferred in context
• Webex Contact Center requires calls to arrive using the G.711 uLaw or G.711 aLaw codec. in context
• Webex Contact Center's PSTN media and signalling requirements list RFC2833 as the DTMF method for calls arriving at Webex Contact Center. in context
• Webex Contact Center specifies a 20 ms packetization timer in the SIP SDP for PSTN ingress. in context
• Webex Contact Center supports SIP Delayed Offer and does not support SIP Early Offer on PSTN ingress. in context
• An entry point is the association of an inbound telephone number with an IVR service flow in Webex Contact Center. in context
• When creating an inbound telephony entry point, the administrator selects a Webex Calling location and a PSTN number from drop-down lists before saving. in context
• Numbers for the contact center are added in Control Hub under Locations, selecting a location, then Calling, Set up calling, choosing a PSTN provider type and using Add Numbers Now. in context
• The PSTN provider types offered for a contact center location are Webex Contact Center PSTN, Cloud Connected PSTN, Local Gateway and Cisco Calling Plan. in context
• Cisco advises successfully making inbound and outbound calls over the chosen Webex Calling PSTN connectivity option before deploying Webex Contact Center. in context
• A valid Webex Calling subscription is required for agents who use Webex devices or the Webex App. in context
• Cloud Connected PSTN providers are provisioned to support +E.164 format numbers for both DID and toll-free numbers used with Webex Contact Center. in context
• Webex Contact Center PSTN media requires G.711 u-law or A-law, and any other codec such as G.729, iLBC or Opus must be transcoded before reaching Webex Contact Center. in context
• Webex Contact Center PSTN requirements specify RFC 2833 DTMF and a 20 ms packetization time in the SDP. in context
• Webex Contact Center PSTN signaling requirements list SIP Early Offer as not supported and SIP Delayed Offer as supported. in context
• The Webex Calling based voice channel setup article states that an inbound entry point requires a Webex Calling location and a PSTN region, and can be populated with a PSTN number, an extension number, or both. disputed in context
• When entry points use extensions Cisco says to avoid mixing prefixed and non-prefixed locations and to avoid overlapping extensions across locations because both can cause unexpected call routing. in context
• Cisco warns administrators to configure only the numbers allocated to their Webex Contact Center PSTN connection because entering invalid numbers may result in call routing failures. in context
• Premises-based PSTN DIDs or toll-free numbers configured in Control Hub for Local Gateway are saved in +E.164 format. in context
• Cisco advises limiting the number of PSTN carriers involved in routing to no more than two before the call reaches Webex Contact Center to keep latency low. in context
• Cisco does not support endpoint devices in the same Webex Calling location as Webex Contact Center PSTN and warns this may cause unexpected overage charges. in context
• Webex Contact Center does not provide transcoding or media termination point services for calls using unsupported codecs. in context
• Webex Contact Center voice supports four PSTN options: Webex Contact Center PSTN, Webex Calling Cloud Connected PSTN, Cisco Calling Plan, and Webex Calling Local Gateway. in context
• Introducing a second upstream PSTN carrier that forwards calls to an intermediary carrier before Webex Contact Center is not a supported configuration for Cloud Connected PSTN, Local Gateway or Webex Contact Center PSTN. in context
• Cisco's Webex Calling configuration checklist for Webex Contact Center says to confirm under Contact Center > Settings > Service Details that the voice media platform is Real Time Media Service and telephony is Webex Calling. in context
• Cisco says to make successful inbound and outbound calls over the chosen Webex Calling PSTN option before completing the remaining Webex Contact Center deployment tasks. in context
• Webex Contact Center PSTN is available in the contiguous United States and Canada only. in context
• The Cisco-provided Webex Contact Center PSTN service does not support international dialing; it delivers calls only to and from North American Numbering Plan destinations in the contiguous 48 US states or Canada. in context
• On a tenant whose PSTN is the Cisco-provided Webex Contact Center PSTN bundle, an outdial to a destination outside the contiguous US and Canada is expected to fail regardless of dial plan or desktop profile settings. inferred in context
• All PSTN numbers (direct-inward-dial or toll-free) are stored in Control Hub in +E.164 format. in context
• When agents log in to Webex Contact Center, they choose a preferred call destination of 1 of 3 types: Dial Number (a free-form DID-style number, typically +E.164 but not mandatory), Extension (a short internal number matched to a Webex Calling/UC endpoint), or Desktop (WebRTC calling in the browser). in context
• Cisco Webex Calling Plan as a Contact Center PSTN option is currently available only for Webex Contact Center trials, and otherwise only in regions where Cisco supports service and/or inbound toll-free numbers. in context
• Webex Cloud Connected PSTN (CCP, also called Cloud Connect or CCCP) provides PSTN connectivity in over 65 countries through Certified Calling Providers. in context
• Using Cloud Connected PSTN requires the customer to partner directly with a Cloud Connected PSTN (CCP) provider, who handles service initiation/activation, billing, and phone number allocation. in context
• Webex Contact Center and Webex Calling share a global network of Common Edge access services, which lets calls route into Webex Contact Center using Webex Calling PSTN services without requiring a Webex Calling subscription. in context
• Webex Contact Center PSTN does not support international dialing; it only delivers calls to/from North American Numbering Plan destinations in the lower 48 states or Canada. in context
• Webex Contact Center PSTN (the Cisco-billed PSTN bundle) is available only in the contiguous United States and Canada. in context
• When an entry point is configured with both an E.164 number and an extension, Webex Contact Center always reports the call against the E.164 number even if the extension was actually dialed. in context
• An Entry Point is the configuration object that associates an inbound telephone number with an IVR service flow (a Webex CC Flow) in Webex Contact Center; inbound numbers must be mapped to entry points as part of PSTN provisioning. in context
• Agents can connect to Webex Contact Center via 5 methods: Webex PSTN inter-connectivity (PSTN DID or cellular agents), Webex Calling registered endpoints (MPP devices or Webex App, requiring a valid Webex Calling subscription), WebRTC Agent Desktops, premise-based PBX/PSTN agents, or Microsoft Teams agents via Local Gateway Direct SIP Routing. in context
• Webex Contact Center administrators can choose from 3 Webex Cloud PSTN interconnectivity options: Webex Contact Center PSTN, Webex Cloud Connected PSTN, and Cisco Calling Plan, plus an on-premises option, Webex Calling Local Gateway. in context
• Calls arriving at Webex Contact Center must use codec G.711uLaw or G.711aLaw with DTMF via RFC2833; any other codec (G.729, iLBC, Opus, etc.) must be transcoded externally before reaching Webex Contact Center, since Webex Contact Center provides no transcoding or media termination point service. in context
• Webex Calling Local Gateway lets a customer keep its existing on-premises PSTN carrier while providing access in over 140 markets globally. in context
• Local Gateway deployments for Webex Contact Center requiring more than 250 concurrent calls (IVR plus agent calls combined) must use a Certificate Based trunk type, which encrypts SIP and RTP toward the Webex Global Media network; a Registration Based trunk is used for up to 250 concurrent calls. in context
• The Local Gateway is the customer's own Session Border Controller (SBC), which registers with Webex Calling and routes calls between Webex Calling and on-premises resources such as agents or existing PSTN connectivity. in context
• For call quality, Webex Contact Center supports at most a single upstream/northbound PSTN carrier forwarding calls into the connected PSTN service (Cloud Connected PSTN, Local Gateway, or Webex Contact Center PSTN); forwarding through a second intermediary carrier before reaching Webex Contact Center is explicitly unsupported due to added latency. in context
• All new Webex Contact Center subscriptions automatically include Webex Calling PSTN services needed for Common Edge; tenants provisioned before October 2024 may need their partner to add Webex Calling PSTN service to use Common Edge services. in context
• Administrators can assign a 'PSTN Region' to an entry point's telephone number in Control Hub, which determines where in Webex's global edge network the contact center processes that call's voice media, to minimize latency for geographically distributed deployments. in context
• Webex Contact Center's SIP interoperability requirements state SIP Early Offer is not supported (listed as a future roadmap item), SIP Delayed Offer is supported, and SIP Early Media is not supported (also a future roadmap item); the SDP packetization timer must be 20ms. in context
• The 2 help.webex.com voice-channel setup URLs (article IDs nhy3codb and 2dputx) serve byte-for-byte identical article HTML/body content; they are the same document published under 2 different article IDs, not a redirect and not distinct content. in context
• Cisco does not recommend enabling voicemail on an agent line, because an incorrectly configured setup could let a customer call go to voicemail while the contact center believes the call was answered normally. in context
• WarmTransfer infers that Cisco has withdrawn vPOP as a documented Webex Contact Center PSTN option, because the onboarding guide has been expired and the current voice-channel article's change log records removing its vPOP reference. No end-of-sale or end-of-service notice was found to confirm this. inferred in context
• Cisco's change log for the current Webex Contact Center voice-channel article records a 25 June 2026 edit titled 'Removed reference to VPoP', and the separate vPOP Onboarding Guide article no longer serves content. in context
• WebRTC agent calling requires a minimum network bandwidth of 100kbps per call. in context
• WebRTC Desktop support for agents is currently unavailable for Webex Contact Center customers hosted in the Japan data center. in context

Cite this source record

APA

WarmTransfer. (2026, June 25). Set up voice channels for Webex Contact Center. WarmTransfer. https://warmtransfer.net/knowledge/sources/cisco-help-nhy3codb-wxcc-voice-channel

BibTeX

@misc{warmtransfer-cisco-help-nhy3codb-wxcc-voice-channel,
  title  = {Set up voice channels for Webex Contact Center},
  author = {{WarmTransfer}},
  year   = {2026},
  url    = {https://warmtransfer.net/knowledge/sources/cisco-help-nhy3codb-wxcc-voice-channel},
  note   = {Cisco Systems, Inc. (Webex Help Center), accessed 2026-09-04}
}