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WebRTC, Call API V2 and Other Improvements

1. Audio Infrastructure Update

We have upgraded our audio infrastructure to WebRTC, moving away from the original websocket-based system. This change ensures better scalability and reliability:

  • Web Calls: All web calls are now on WebRTC.
  • Phone Calls: Migration to WebRTC is in progress, pending resolution of some SIP blockers.

2. Call API V2

We've introduced the updates in our Call API V2, which now separates phone call and web call objects and includes a few field and API changes:

See Doc

3. Concurrency Enhancements

  • Default Limits: The default concurrency limit for all users has been increased to 20.
  • Concurrency API: A new API to check your current concurrency and limit is now available here.

4.  Seperation inbound and outbound

  • Agent Separation: Our APIs now support separate inbound and outbound agents, with the option to disable either as needed.
  • Nickname Field: Easily find specific numbers with the addition of a nickname field for better organization.

5. Bug Fix and Reliability Improvement

  • Enhanced all modules with a smarter retry and failover mechanism.
  • Resolved issues with audio choppiness and looping.
  • Corrected the display of function call results in the LLM playground.
  • Addressed the scrolling issue in the history tab.

6. Usage Limits

In response to abuse and misuse of our platform, we added some usage limits accordingly:

  • Scam Detection: Implemented to safeguard users.
  • Call Length Limit: Maximum of 1 hour.
  • Token Length Limit: Maximum of 8192 tokens for Retell LLM. For multi-state prompts, this includes the longest state plus the general prompt.
  • Please contact us if you need exceptions.