w3c/webrtc-rtptransport
Repository for the RTPTransport specification of the WebRTC Working Group
HTMLNOASSERTION
Issues
- 3
- 0
BYOB also needs fields giving the byte length
#62 opened by tonyherre - 0
Allow inserting padding into packets
#60 opened by tonyherre - 11
Make RtpTransportProcessor transferable
#33 opened by pthatcher - 2
RtpSendStreamInit and RtpReceiveStreamInit are not defined in explainer-use-case-1.md
#35 opened by youennf - 0
- 0
No signal when transport path change.
#57 opened by Philipel-WebRTC - 1
Sending NACK/FEC is not supported
#52 opened by Philipel-WebRTC - 9
RtpTransport should be called RTCRtpTransport
#2 opened by henbos - 0
Potential issues from convenience extensions
#39 opened by aboba - 6
- 1
Controls of internal queue lengths
#54 opened by tonyherre - 0
- 11
Custom extensions not supported
#29 opened by Philipel-WebRTC - 7
- 6
Sequence number: constraints?
#43 opened by aboba - 2
BYOB interfaces to avoid ArrayBuffer churn
#41 opened by tonyherre - 8
Not requiring per-packet JS Events
#20 opened by tonyherre - 0
- 1
RtpTransport and transport re-direction
#48 opened by aboba - 4
- 1
Only allow RtpTransport (max-bundle)
#34 opened by pthatcher - 2
Packet prioritization is not supported.
#38 opened by Philipel-WebRTC - 2
Packet forwarding is not supported
#40 opened by Philipel-WebRTC - 0
Clarify RTCP use of Use Case 1
#27 opened by pthatcher - 2
Arbitrary RTP Header Extensions
#12 opened by aboba - 0
Use case 1: NACK/RTX control and RTCP
#26 opened by aboba - 3
- 2
- 1
Update examples to use standard APIs
#14 opened by jan-ivar - 1
Arbitrary RTCP messages
#11 opened by aboba - 1
WHATWG Streams support?
#8 opened by aboba - 4
How does cryptex work?
#7 opened by aboba - 3
SDP "Bumper lanes"
#10 opened by aboba - 1
Customizing piecemeal
#9 opened by aboba