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7.1.9.9604591078 - WebRTC Voice Friday, June 21, 2024 Release Notes WebRTC Voice WebRTC (Web Real-Time Communications) is the predominant telephony protocol used by web-based applications, such as Google Meet. It’s built-in to Chrome, Safari, Firefox, and many other web browsers, and allows voice communication as well as data and video.
This project adds WebRTC as the primary voice provider for SecondLife, replacing the dated Vivox-based voice system we’ve had for a decade and a half.
Currently WebRTC voice may only be experienced in the region: WebRTC Voice 1 (more being added) Please see our blog post for more details https://community.secondlife.com/blogs/entry/15626-coming-to-an-agni-region-near-you-an-update-on-changes-to-voice/
Features provided by WebRTC include: Improved NAT hole punching. Audio/video device selection. Improved spatialization. 48khz audio bandwidth providing cleaner sound. User control over audio noise reduction - high reduction for noisy environments, no reduction for clean audio sources (performers, etc.) User control over automatic gain control - less need to individually tune other user’s audio levels. Improved audio echo cancellation. Improved security - person-to-person communication goes through our servers, and won’t leak personal details such as IP addresses. And many more. Most of the same features are still available:
Spatial in-world voice. Peer to peer sessions. Voice conferences with multiple others. Group voice (including moderation) Feature changes:
Voice morphing is not included with WebRTC. (There are many apps that provide these services.) Conferences and group voice calls are limited to 50 participants. Known issues: https://github.com/secondlife/viewer/issues/1806 Repeatedly starting ad hoc calls may crash your viewer.You can’t make P2P calls from WebRTC enabled viewers to vivox-only viewers. Group and conference calls default to WebRTC or Vivox depending on whether the call was initiated by someone running the webrtc viewer. When establishing a p2p/group/conference call with WebRTC and then TPing to a Vivox region, the voice drops. On some systems, there is an occasional ‘stutter’ in frame rate when crossing parcel or region boundaries. The “Speak” button in the conversation floater is displayed as disabled after connecting to the IM, Ad-hoc, and Group chats. If you do have issues with voice, disable and re-enable voice. Resolved Issues viewer#1090 [WebRTC] Voice input/output is not using the Windows-Default device when set to 'Default'viewer#1732 Crash after toggling Voice Echo Cancellation off and onviewer#1776 Viewer crash when connecting to webRTC voice with certain settings.xmlviewer#1788 Disabling the Media preference should not disable the Noise Suppression preferenceviewer-private#115 WebRTC - logging - Very little webrtc logging around voice state