WebRTC Virtual Meeting Decisions Summary


 Impact on my application

 Standardization status



Once in a while, between IETF meetings, there are virtual meeting held to speed up standardization work.  In last month’s teleconference, several decisions were made.  In this post we are listing some of the most important decisions made:

Transceiver lifetime

Stopping a received track will not stop the associated transceiver, which will continue sending RTCP.

New voice activity detection control

A new API control to enable/disable voice activity detection will be added.  We will do a short future post on this.

Codec specificity increased

Codec controls will now use both MIME media type and subtype.  We will do a short future post on this.

RTCRtpSender/Receiver created synchronously before setLocal/Remote

addTransceiver() and addTrack() will cause RTCRtpSenders/Receivers to be created (synchronously) even though their transports will not exist until after setLocal/RemoteDescription.

Reuse track in addTranceiver()

It will (continue to) be permitted to supply the same track to addTransceiver() more than once.

addStream and removeStream are out

After much discussion, it was confirmed that legacy stream-based methods such as addStream and removeStream will remain out of the specification.


 Impact on my application

Immediate impact as these decisions were made and will find their way into the implementation


 Standardization status