# Blog post on SIP dial-out and cascaded mixing in the AudioBridge

**URL:** <https://janus.discourse.group/t/blog-post-on-sip-dial-out-and-cascaded-mixing-in-the-audiobridge/1585>\
**Category:** Announcements\
**Created:** [April 3, 2025, 3:36pm UTC](https://janus.discourse.group/t/blog-post-on-sip-dial-out-and-cascaded-mixing-in-the-audiobridge/1585 "2025-04-03T15:36:25Z")\
**Posts on this page:** 1\
**Page:** 1

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**Author:** ![lorenzo](https://yyz2.discourse-cdn.com/free1/user_avatar/janus.discourse.group/lorenzo/32/425_2.png) [@lorenzo](https://janus.discourse.group/u/lorenzo)\
**Post date:** [April 3, 2025, 3:36pm UTC](https://janus.discourse.group/t/blog-post-on-sip-dial-out-and-cascaded-mixing-in-the-audiobridge/1585/1 "2025-04-03T15:36:25Z")

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Enjoy!

> **[Dial-out and cascaded mixing with the AudioBridge | Meetecho Blog](https://www.meetecho.com/blog/cascaded-mixing/)**
>
> A couple of years ago I wrote a blog post explaining how I used some existing features in the AudioBridge plugin to allow SIP endpoints to dial-in and join a conversation that would normally be limited to WebRTC participants alone. That worked nicely...

Lorenzo