[improvement]: Add playback progress acknowledgment for WebSocket media (per-chunk or byte-level acknowledgment)

[gauravs456]

Is your feature or improvement request related to a problem? Please describe.

Asterisk’s WebSocket media driver (chan_websocket) allows streaming binary audio data from external applications (like AI voice bots) but does not provide any mechanism to know when a specific portion of audio has actually been played.

This causes synchronization issues for AI-driven real-time streaming systems that generate audio dynamically (e.g., Text-to-Speech or conversational AI). Without playback progress acknowledgment, the application has no way to determine when the audio it sent has finished playing.

This leads to two major problems:

1. Over-buffering — increases latency since the application keeps sending new audio before the previous one is played.
2. Under-buffering — causes playback gaps when the application waits too long to send the next chunk.

Describe the solution you'd like

Introduce a playback progress acknowledgment mechanism for the WebSocket media driver.

Possible designs:

1. Mark-based acknowledgment

   • Allow clients to send a MARK id=<uuid> text control command that sets a logical boundary in the playback queue.
   • When Asterisk finishes playing all media queued before that mark, it responds with MARK_PLAYED id=<uuid>.

   Example:

Describe alternatives you've considered

We explored existing Asterisk WebSocket control messages:

• REPORT_QUEUE_DRAINED: Notifies only when the entire queue is empty — not useful for partial playback acknowledgment.
• FLUSH_MEDIA: Clears buffered audio but provides no confirmation of playback.
• MEDIA_XOFF / MEDIA_XON: Flow control only, unrelated to playback timing.

We also attempted to simulate playback timing locally by estimating real-time audio consumption (1 ms per 8 bytes for μ-law), but this approach is only an approximation and cannot confirm actual playback progress in Asterisk.

Additional context

Use case: AI-driven voice bots and real-time speech generation systems streaming audio to Asterisk over WebSocket.

Modern AI engines (like OpenAI Realtime API, ElevenLabs, or custom TTS models) generate audio in small, variable-sized chunks. These systems require acknowledgment when certain chunks have been played so they can dynamically:

• Generate the next segment of speech
• Handle interruptions or “barge-in” events
• Avoid excessive buffering and latency

Adding per-chunk or progress-based acknowledgment would significantly improve synchronization for real-time applications and make Asterisk more compatible with emerging AI voice technologies.

Proposed area: WebSocket Media Driver (chan_websocket)

[gtjoseph]

@gauravs456 Do you have time to submit a PR for this yourself?

[gauravs456]

Hi @gtjoseph i can try but have one doubt you have done a work for support of passing websocket events in json format ,so for this work if i fork master then do the changes then again for this work json event need to implemented so should i fork from your branch or
i can do it from master?? with json supported ??

[gtjoseph]

The PR with the JSON format will be merged today so if you wait until tomorrow, you can just use the "master" branch as the base. You'll only have to support JSON going forward.

[gauravs456]

Hi @gtjoseph, i am sorry but i am bit occupied this month due to critical releases in work and i checked the code in master there will be some major changes that need to be done
do you mind if you can cop up this task?

[gtjoseph]

No worries. We'll add it to our backlog.

[gtjoseph]

@gauravs456 Give PR #1577 a test please.