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BEGIN:VEVENT
UID:71701032-4e1f-463a-b875-f6608345645a
DTSTAMP:20241016T223346Z
SUMMARY:RtpTransport
DTSTART;TZID=America/Los_Angeles:20240925T100000
DTEND;TZID=America/Los_Angeles:20240925T110000
DESCRIPTION:https://www.w3.org/events/meetings/71701032-4e1f-463a-b875-f660
 8345645a/\n\nWhile [WebCodecs ](https://w3c.github.io/webcodecs/ )provides
  low-level access to the browser's native encoders and decoders\, today th
 ere is no API to transport media peer-to-peer using the Real-time Transpor
 t Protocol ([RTP](https://en.wikipedia.org/wiki/Real-time_Transport_Protoc
 ol))\, defined in [RFC 3550](https://www.rfc-editor.org/rfc/rfc3550).\n\nT
 he goal of the [RtpTransport API](https://github.com/w3c/webrtc-rtptranspo
 rt) is to enable applications to utilize peer-to-peer RTP transport\, as w
 ell as to send and respond to feedback using the Real-time Control Protoco
 l (RTCP).\n\nThe following use cases are being developed:\n\n- [Use case 1
 : Custom Packetization](https://github.com/w3c/webrtc-rtptransport/blob/ma
 in/explainer-use-case-1.md)\n- [Use case 2: Custom Congestion Control](htt
 ps://github.com/w3c/webrtc-rtptransport/blob/main/explainer-use-case-2.md)
  \n- [Use case 3: Custom NACK/RTX](https://github.com/w3c/webrtc-rtptransp
 ort/blob/main/explainer-use-case-3.md)\n\nSee:\n- [GitHub repo](https://gi
 thub.com/w3c/webrtc-rtptransport).\n- [Specification](https://w3c.github.i
 o/webrtc-rtptransport/)\n\nAgenda\n\n**Chairs:**\nBernard Aboba\, Peter Th
 atcher\, Erik Språng\n\n**Description:**\nWhile [WebCodecs ](https://w3c.
 github.io/webcodecs/ )provides low-level access to the browser's native en
 coders and decoders\, today there is no API to transport media peer-to-pee
 r using the Real-time Transport Protocol ([RTP](https://en.wikipedia.org/w
 iki/Real-time_Transport_Protocol))\, defined in [RFC 3550](https://www.rfc
 -editor.org/rfc/rfc3550).\n\nThe goal of the [RtpTransport API](https://gi
 thub.com/w3c/webrtc-rtptransport) is to enable applications to utilize pee
 r-to-peer RTP transport\, as well as to send and respond to feedback using
  the Real-time Control Protocol (RTCP).\n\nThe following use cases are bei
 ng developed:\n\n- [Use case 1: Custom Packetization](https://github.com/w
 3c/webrtc-rtptransport/blob/main/explainer-use-case-1.md)\n- [Use case 2: 
 Custom Congestion Control](https://github.com/w3c/webrtc-rtptransport/blob
 /main/explainer-use-case-2.md) \n- [Use case 3: Custom NACK/RTX](https://g
 ithub.com/w3c/webrtc-rtptransport/blob/main/explainer-use-case-3.md)\n\nSe
 e:\n- [GitHub repo](https://github.com/w3c/webrtc-rtptransport).\n- [Speci
 fication](https://w3c.github.io/webrtc-rtptransport/)\n\n**Goal(s):**\nTo 
 describe the goals of the RtpTransport API\, and to solicit feedback on de
 veloper needs and use cases.\n\n\n**Agenda:**\n1. Introduction to the RtpT
 ransport API.\n2. Use cases.\n3. Developer feedback.\n\n**Materials:**\n- 
 [slides](https://www.w3.org/2024/Talks/TPAC/breakouts/rtp-transport.pdf)\n
 - [minutes](https://www.w3.org/2024/09/25-rtp-transport-minutes.html)\n- [
 Session proposal on GitHub](https://github.com/w3c/tpac2024-breakouts/issu
 es/13)\n\n**Track(s):**\n- Real-time Web
STATUS:CONFIRMED
CREATED:20240916T214822Z
LAST-MODIFIED:20241016T223346Z
SEQUENCE:1
ORGANIZER;CN=W3C Calendar;PARTSTAT=ACCEPTED;ROLE=NON-PARTICIPANT:mailto:nor
 eply@w3.org
LOCATION:-1 Lower Level - Catalina 2
CATEGORIES:TPAC 2024,Breakout Sessions
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