Partners and end customer ask very often about the different possibilities they have to integration WebRTC-based click to call buttons with their existing architecture. Most of them are willing to use the current platforms in order optimise agents time and existing metrics. And this is possible the WebRTC call is adapted to SIP so the legacy ACD can route the voice call to an available agent. This approach has the following benefits:
- Agents can answer calls of other services, WebRTC-based calls are not more than a new channel. No dedicated agents for WebRTC
- ACD, route policies, statistics and KPI are preserved !!
- Reduced cost, no adaptation problems.
- A SIP trunk is setup between Sippo and the PBX for voice only
- Voice is sent from Sippo to the PBX, that routes them to an available agent
- Video is sent from Sippo to the agent’s browser
- Agents use PBXs softphone for voice only, and Sippo WebCollaborator for video and collaboration
- Call correlation is done via and ID sent on the SIP headers and/or shown to the customer
- Media flows from the customer browser to the agent browser
- PBX is only used for querying agent’s availability, without SIP integration
- Agents use Sippo Web Collaborator for audio, video and collaboration.
- A SIP trunk is setup between Sippo and the PBX for voice and video
- Calls are sent from Sippo to the PBX, that routes them to an available agent
- Agents use their existing video softphone for audio and video, while using Sippo for collaboration
- A SIP trunk is setup between Sippo and the PBX for voice and video
- Agents SIP-register into the PBX
- Calls are sent from Sippo to the PBX, that routes them to an available agent
- Agents use Sippo Web Collaborator for voice, video and collaboration
- Call correlation is done via and ID sent on the SIP headers and/or shown to the customer