SIP Mediaserver Service - How It Works

The iotcomms.io SIP Mediaserver Service provides versatile functions for integrating voice and media handling capabilities into your applications via REST APIs. The service is designed to enable easy development of interactive voice and call control applications. Its fine-grained API lets you automatically route calls and design interactive communication flows the way you want, making the service uniquely flexible in use cases such as customer service systems, contact center applications, conference calls, multi-party voice applications, Alexa device communication, and unified communications. The SIP Mediaserver Service acts as a media handler and call controller, while your application determines the call logic through a series of REST callbacks.

The SIP Mediaserver Service is available as a SaaS offer, in the cloud or in a hybrid cloud deployment model. The hybrid cloud deployment option lets you run the service within your own data center using the Hybrid Enabler Service to meet highest privacy standards while the iotcomms.io team manages and monitors the service for best operational efficiency.

The SIP Mediaserver Service allows your applications to manage and interact with voice sessions through REST callbacks and session-control APIs. The same application-driven model is used for two-party calls, IVR flows, and conference mixer sessions where multiple participants are joined into a shared mixed audio connection. The same model is also used when a call participant is an Alexa device connected through the iotcomms.io Alexa Interface. In that case, the service handles the Alexa call signaling and media interworking while the application continues to make call-flow decisions through the SIP Mediaserver callbacks.

1. /newCall: Triggered when a new incoming call is detected, notifies the your application. The application can use this event to decide whether to accept, route, or reject the call.

2. /getIvrCommand: A key callback for interactive voice applications, this endpoint provides instructions for playing prompts, collecting user input (like DTMF tones), or moving forward in the call flow. This callback is integral to creating interactive and responsive user experiences.

3. /callEvent: Triggered on call state changes. Your application's response to callEvent dictates call flow decisions---such as when to bridge participants, join a participant into a conference mixer connection, terminate the call, or invoke the interactive voice response (IVR) function. This allows for flexible call control and real-time call management.

1. Receiving Incoming Calls: When the SIP Mediaserver Service detects a new call, it triggers the /newCall callback, sending the call information to your application. The application can analyze the incoming call data and respond to indicate whether to answer, reject, or forward the call.

2. Managing Call Flow with REST Callbacks: Once a call is answered, your application can guide the interaction using the /getIvrCommand and /callEvent callbacks:

   • /getIvrCommand: This callback enables your application to issue commands for playing audio prompts, text-to-speech phrases, prompting for DTMF input, or specifying any custom actions needed within the call.

   • /callEvent: This callback provides your application with real-time updates on the call status. Based on these updates, the application can decide how to handle the session; whether to connect additional participants, redirect, or end the call. This enables the application to build complex call-handling scenarios tailored to fit user interactions and business requirements.

3. Connecting Call Participants: Through responses to /callEvent, your application can control the connections between call participants, such as bridging a caller with an agent or another party. This function is crucial for use cases requiring dynamic call routing and multi-party interactions, such as in customer service or emergency response scenarios.

4. Conference Mixer Sessions: Through the Conference Mixer functionality, your application can create a shared conference connection and join multiple participants into the same mixed audio session. This enables conference calls and multi-party voice applications where participants can be added, parked with IVR, moved from existing connections, and removed under application control.

5. Alexa-enabled Sessions: When Alexa devices are provisioned as participants, the SIP Mediaserver Service can be used to include the device in the same REST-controlled session model as SIP and WebRTC participants. This enables applications such as hospitality guest services, senior living communication, and healthcare workflows where a user can start a voice session from an Alexa device and be routed to the right service or staff destination.