> For the complete documentation index, see [llms.txt](https://docs.ixhello.com/llms.txt). Markdown versions of documentation pages are available by appending `.md` to page URLs; this page is available as [Markdown](https://docs.ixhello.com/ixhc2/technical-specifications/audio-channel-architecture-and-call-bridge-framework.md).

# Audio Channel Architecture and Call Bridge Framework

### Overview

The platform is evolving toward a more modular and scalable voice architecture through the introduction of a new component called Call Bridge.

Several existing platform services continue to be leveraged, including:

* Speech Services
* Runtime Engine
* AI-Vendors
* Channel Gateway

The new Call Bridge module acts as a central audio integration layer that simplifies the process of connecting new voice channels and telephony providers to the platform.

{% hint style="info" %}
Note:  This integration is only available in demo.  It will be fully available in 2.4 relase
{% endhint %}

***

## Purpose of Call Bridge

Call Bridge consolidates the complex audio-processing and telephony-related functionality into a reusable framework. Instead of each audio integration implementing the entire voice stack independently, channel developers only need to implement a defined set of interfaces.

#### Key Benefits

* Simplified onboarding of new voice channels
* Reduced integration complexity
* Consistent audio processing across providers
* Reusable telephony capabilities
* Easier expansion to future voice platforms

***

## How the Integration Works

#### High-Level Call Flow

* A customer call enters a Genesys Architect flow.
* Architect invokes the Call Audio Connector action.
* Genesys establishes a secure WebSocket connection with VNext.
* Audio is streamed bi-directionally between Genesys and VNext.
* VNext processes the conversation using:
  * Speech-to-Text (STT)
  * Natural Language Understanding (NLU)
  * Text-to-Speech (TTS)
* VNext captures customer intent and relevant business information.
* When the conversation ends, VNext returns the configured output variables to Genesys.
* Architect resumes execution and routes the call to the appropriate agent queue or workflow.

<figure><img src="https://4187494610-files.gitbook.io/~/files/v0/b/gitbook-x-prod.appspot.com/o/spaces%2FFe4cpY1jy0FJ4kgC7wSx%2Fuploads%2FINrtBmrpLuzxMnGO7Jqz%2Fimage.png?alt=media&amp;token=da515e60-8582-4396-9726-ddb54243f8ff" alt=""><figcaption></figcaption></figure>

***

## Scalability Improvements

One of the primary goals of the new architecture is to improve scalability.

#### Current Challenge

Previous implementations faced limitations such as port restrictions that constrained environment size.

#### Expected Improvement

With the introduction of FreeSWITCH and the Call Bridge architecture:

* Traditional port limitations can be removed
* Environment capacity becomes primarily dependent on available infrastructure resources
* Larger-scale deployments become possible

***

## Audio Connector Framework

To integrate a new audio channel, developers must implement a standardized **Phone Interface**.

The interface allows an audio connector to advertise its supported capabilities to the Call Bridge framework.

<figure><img src="https://4187494610-files.gitbook.io/~/files/v0/b/gitbook-x-prod.appspot.com/o/spaces%2FFe4cpY1jy0FJ4kgC7wSx%2Fuploads%2FO9FgscA5ozjAHoCPXfmW%2Fimage.png?alt=media&amp;token=ccf1ff1a-07c6-4a96-9c01-8448587eeb8f" alt=""><figcaption></figcaption></figure>

### Configure a Genesys Audio Connector Connection

Refer to the [set-up document](/ixhc2/integrations/genesys-integration-setup-with-ix-hello-customer-v2.md)

### Advertised Capabilities

An audio connector can specify support for:

* Audio format support
* Sample rate configuration
* Outbound calling
* Call transfer
* Call recording
* DTMF detection
* DTMF playback
* Audio playout tracking

These capabilities enable the platform to understand what functionality is available for a given voice channel.

***

## Recommended Audio Standard

The platform is moving toward standardization on:

**L16 Audio Format**

Characteristics:

* Linear PCM audio
* 16-bit audio format
* Commonly used as a universal audio processing format within the platform

Standardizing on L16 reduces channel-specific implementation complexity and enables consistent speech processing.

***

## Call Lifecycle Management

Once a voice interaction is established, the connector is responsible for creating a Call Object.

The Call Object provides:

#### Events

* Inbound call events
* Outbound call events
* Call state updates
* Audio stream events

#### Lifecycle Handling

Developers must correctly implement the required event handling and lifecycle behaviors to ensure reliable call processing.

After integration is complete, the connector gains access to platform voice capabilities, including:

* Speech Recognition
* Speech Synthesis
* Real-time audio processing
* Runtime Engine integrations
* AI-driven conversational experiences

<figure><img src="https://4187494610-files.gitbook.io/~/files/v0/b/gitbook-x-prod.appspot.com/o/spaces%2FFe4cpY1jy0FJ4kgC7wSx%2Fuploads%2FUNuo9yBirVUTQSTPwuhi%2Fimage.png?alt=media&amp;token=67c57056-abb6-4480-9068-ae92fe7eb579" alt=""><figcaption></figcaption></figure>

***

## Audio Transcoding

Different voice providers and telephony networks use different codecs. To support interoperability, Call Bridge includes built-in transcoding capabilities.

### Included Codec Support

The framework contains codec resamplers that can handle:

* Sample-rate conversion
  * Example: 24 kHz → 8 kHz
* μ-law (Mu-Law)
* A-law

Developers may also:

* Use their own codec libraries
* Integrate third-party audio-processing libraries when needed

***

## PSTN Compatibility

For Public Switched Telephone Network (PSTN) connectivity, audio may need to be transcoded from the platform's internal format (L16 PCM) to telephony codecs such as:

* G.711
* Other provider-specific formats

The Call Bridge framework provides the necessary foundation for these conversions while allowing extensions when additional codecs are required.

***

## WhatsApp Voice Codec Considerations

Discussion during the session highlighted that WhatsApp voice services may support codecs such as:

* Opus
* AMR-related codecs
* Other low-bandwidth voice formats

If additional codec support becomes necessary, the framework can be extended through supplementary codec libraries.

***

## DTMF Support Requirements

A critical requirement for audio-channel implementations is proper DTMF handling.

#### Supported Scenarios

* Receiving DTMF tones
* Sending DTMF tones
* IVR navigation
* External telephony integrations

#### Important Consideration

Any configured codec must support reliable in-band DTMF transmission when required.

This is especially important when calls traverse:

* PSTN networks
* IVR systems
* Third-party telephony platforms

Failure to support DTMF correctly may impact menu navigation and call-routing functionality.

***

## Architectural Benefits

The introduction of Call Bridge provides significant advantages:

Simplified voice-channel development

Reusable telephony infrastructure

Reduced implementation effort

Consistent audio-processing pipeline

Scalable deployment model

Easier support for future voice providers

Standardized codec and audio handling

Direct access to platform speech and AI capabilities

***

## Conclusion

The Call Bridge architecture serves as a unified telephony framework that abstracts the complexities of audio processing, codec management, call lifecycle handling, and channel integration. By implementing a small set of standardized interfaces, new voice channels can quickly leverage the platform's existing speech services, runtime engine, and AI capabilities, accelerating development while improving scalability, maintainability, and future extensibility.


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