Cloudflare changelogs | Realtime

Realtime - Post-meeting transcriptions are now Generally Available in RealtimeKit

Mon, 08 Jun 2026 00:00:00 GMT

RealtimeKit lets you build products where people meet over live audio and video — such as HealthTech, EdTech, proctoring, and other real-time platforms — on Cloudflare's global WebRTC infrastructure.

Post-meeting transcription is now Generally Available, so completed RealtimeKit meetings can automatically produce full transcript files after they end. Those transcripts can also power AI-generated summaries for meeting notes, review workflows, and follow-up tasks after the transcript is available.

Post-meeting transcription is a managed service powered by Workers AI using Whisper Large v3 Turbo. RealtimeKit handles transcription processing and can return transcript and summary files through webhooks or the REST API, so you do not need to run your own transcription infrastructure.

Generate transcripts and summaries

To generate a transcript after a meeting ends, set transcribe_on_end: true when creating a meeting. To also generate an AI summary automatically after the transcript is available, set summarize_on_end: true:

curl -X POST "https://api.cloudflare.com/client/v4/accounts/$ACCOUNT_ID/realtime/kit/$APP_ID/meetings" \
  -H "Authorization: Bearer $CLOUDFLARE_API_TOKEN" \
  -H "Content-Type: application/json" \
  -d '{
    "title": "Weekly product review",
    "transcribe_on_end": true,
    "summarize_on_end": true,
    "ai_config": {
      "transcription": {
        "language": "en"
      },
      "summarization": {
        "word_limit": 500,
        "text_format": "markdown",
        "summary_type": "team_meeting"
      }
    }
  }'

Consume results

When RealtimeKit finishes processing a meeting, it creates download URLs for the transcript and, if summarize_on_end is set, the summary. You can receive those URLs automatically with webhooks, or fetch them later for a specific session with the REST API.

To receive results as soon as they are ready, configure the meeting.transcript and meeting.summary webhook events:

curl -X POST "https://api.cloudflare.com/client/v4/accounts/$ACCOUNT_ID/realtime/kit/$APP_ID/webhooks" \
  -H "Authorization: Bearer $CLOUDFLARE_API_TOKEN" \
  -H "Content-Type: application/json" \
  -d '{
    "name": "AI results webhook",
    "url": "https://example.com/webhook",
    "events": ["meeting.transcript", "meeting.summary"],
    "enabled": true
  }'

To fetch results later, call the transcript or summary endpoint for the session:

curl -X GET "https://api.cloudflare.com/client/v4/accounts/$ACCOUNT_ID/realtime/kit/$APP_ID/sessions/$SESSION_ID/transcript" \
  -H "Authorization: Bearer $CLOUDFLARE_API_TOKEN"

curl -X GET "https://api.cloudflare.com/client/v4/accounts/$ACCOUNT_ID/realtime/kit/$APP_ID/sessions/$SESSION_ID/summary" \
  -H "Authorization: Bearer $CLOUDFLARE_API_TOKEN"

Use the Generate summary of transcripts for the session API only if summarize_on_end was not set and you want to generate a summary manually after the transcript is available:

curl -X POST "https://api.cloudflare.com/client/v4/accounts/$ACCOUNT_ID/realtime/kit/$APP_ID/sessions/$SESSION_ID/summary" \
  -H "Authorization: Bearer $CLOUDFLARE_API_TOKEN"

Post-meeting transcription supports CSV, JSON, SRT, and VTT transcript outputs, automatic language detection and Whisper language codes. RealtimeKit also supports real-time transcription with Deepgram Nova-3 for live captions, in-meeting accessibility, and real-time note-taking.

Learn more in the RealtimeKit transcription docs and summary docs.

Realtime - Cloudflare's Realtime WebSocket adapter now auto-reconnects and buffers WebRTC media

Fri, 29 May 2026 00:00:00 GMT

Cloudflare Realtime SFU is a WebRTC Selective Forwarding Unit that runs on Cloudflare's global network, so you can route live audio, video, and data between WebRTC clients around the world without managing SFU infrastructure or regions.

When you use the WebSocket adapter to stream WebRTC media to a WebSocket endpoint, the adapter now auto-reconnects and buffers audio and video after brief endpoint disconnects or restarts.

Streaming WebRTC media to WebSocket endpoints

Many teams also use Realtime SFU as the media layer for backend applications, such as transcription, recording, note-taking, and agentic media-processing services. These systems often need to consume live WebRTC audio or video from the SFU in backend infrastructure, including Durable Objects, Workers, Containers, or external services, without running a WebRTC client themselves.

The WebSocket adapter bridges that gap by streaming WebRTC media from the SFU to a standard WebSocket endpoint as application-consumable payloads: PCM audio frames and JPEG video frames.

What changed

When you use the WebSocket adapter in Stream mode (egress) to send live audio or video from the SFU to your own WebSocket endpoint, the SFU now automatically reconnects after brief endpoint disconnects or restarts. This is especially helpful for long-running media pipelines where the WebSocket endpoint may briefly restart while a recording, transcription, or live analysis job is still in progress.

Previously, a brief disconnect from your WebSocket endpoint could close the adapter and require your application to recreate it before media could resume. Now, the SFU retries the same endpoint for up to 5 seconds with no API change required. If the endpoint comes back within that window, audio and video delivery resumes automatically.

The reconnect behavior also includes live-first media buffering, so brief interruptions reduce media loss without replaying stale video.

Reconnect behavior

During reconnect:

Audio uses a short bounded backlog to reduce audible loss. If the interruption lasts longer than the backlog can cover, older audio may be dropped.
Video resumes from the latest available JPEG frame instead of replaying stale frames.
Recovery is best effort and does not guarantee gapless or exactly-once delivery.

If the endpoint remains unavailable after the 5-second reconnect window, the adapter closes and must be recreated.

Learn more

Realtime - Record specific participant audio tracks in RealtimeKit

Thu, 28 May 2026 00:00:00 GMT

You can now record specific participant audio tracks in RealtimeKit with track recording. Track recording creates separate WebM files for each participant instead of a single composite recording, which is useful for post-processing, transcription, and regulated or content-sensitive workflows.

To record specific participants, pass user_ids when starting a track recording:

curl --request POST \
  --url https://api.cloudflare.com/client/v4/accounts/<account_id>/realtime/kit/<app_id>/recordings/track \
  --header 'Authorization: Bearer <api_token>' \
  --header 'Content-Type: application/json' \
  --data '{
  "meeting_id": "97440c6a-140b-40a9-9499-b23fd7a3868a",
  "user_ids": ["user-123", "user-456"]
}'

To pass user_ids for selective track recording, use the following minimum SDK versions:

Web Core: @cloudflare/realtimekit version 1.4.0 or later
Web UI Kit: @cloudflare/realtimekit-ui, @cloudflare/realtimekit-react-ui, or @cloudflare/realtimekit-angular-ui version 1.1.2 or later
Android Core or iOS Core: version 2.0.0 or later
Android UI Kit or iOS UI Kit: version 1.1.0 or later

RealtimeKit provides SDKs and UI components so that you can build your own meeting experience on Cloudflare's global WebRTC infrastructure. Teams today build products ranging from telehealth to education on RealtimeKit for global audiences. You can get started today with our Quickstart or take a look at our Cloudflare Meet repo as a reference.

Workers AI, Realtime - Real-time transcription in RealtimeKit now supports 10 languages with regional variants

Fri, 06 Mar 2026 00:00:00 GMT

Real-time transcription in RealtimeKit now supports 10 languages with regional variants, powered by Deepgram Nova-3 running on Workers AI.

During a meeting, participant audio is routed through AI Gateway to Nova-3 on Workers AI — so transcription runs on Cloudflare's network end-to-end, reducing latency compared to routing through external speech-to-text services.

Set the language when creating a meeting via ai_config.transcription.language:

{
  "ai_config": {
    "transcription": {
      "language": "fr"
    }
  }
}

Supported languages include English, Spanish, French, German, Hindi, Russian, Portuguese, Japanese, Italian, and Dutch — with regional variants like en-AU, en-GB, en-IN, en-NZ, es-419, fr-CA, de-CH, pt-BR, and pt-PT. Use multi for automatic multilingual detection.

If you are building voice agents or real-time translation workflows, your agent can now transcribe in the caller's language natively — no extra services or routing logic needed.