-
Notifications
You must be signed in to change notification settings - Fork 10
Commit
This commit does not belong to any branch on this repository, and may belong to a fork outside of the repository.
Spelling fixes and fix url to latest draft (#19)
* fix url to latest draft * spelling fixes * readme spelling fixes * more spelling fixes
- Loading branch information
1 parent
f97dfa0
commit ff074fd
Showing
2 changed files
with
16 additions
and
16 deletions.
There are no files selected for viewing
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters
Original file line number | Diff line number | Diff line change |
---|---|---|
@@ -1,15 +1,15 @@ | ||
# WHIP - WebRTC HTTP ingest protocol draft | ||
|
||
While WebRTC has been very sucessful in a wide range of scenarios, its adoption in the broadcasting/streaming industry is lagging behind. | ||
While WebRTC has been very successful in a wide range of scenarios, its adoption in the broadcasting/streaming industry is lagging behind. | ||
Currently there is no standard protocol (like SIP or RTSP) designed for ingesting media into a streaming service using WebRTC and so content providers still rely heavily on protocols like RTMP for it. | ||
|
||
These protocols are much older than WebRTC and by default lack some important security and resilience features provided by WebRTC with minimal overhead and additional latency. | ||
|
||
The media codecs used for ingestion in older protocols tend to be limited and not negotiated. WebRTC includes support for negotiation of codecs, potentially alleviating transcoding on the ingest node (wich can introduce delay and degrade media quality). Server side transcoding that has traditionally been done to present multiple renditions in Adaptive Bit Rate Streaming (ABR) implementations can be replaced with [simulcasting](https://webrtcglossary.com/simulcast/) and SVC codecs that are well supported by WebRTC clients. In addition, WebRTC clients can adjust client-side encoding parameters based on RTCP feedback to maximize encoding quality. | ||
The media codecs used for ingestion in older protocols tend to be limited and not negotiated. WebRTC includes support for negotiation of codecs, potentially alleviating transcoding on the ingest node (which can introduce delay and degrade media quality). Server side transcoding that has traditionally been done to present multiple renditions in Adaptive Bit Rate Streaming (ABR) implementations can be replaced with [simulcasting](https://webrtcglossary.com/simulcast/) and SVC codecs that are well supported by WebRTC clients. In addition, WebRTC clients can adjust client-side encoding parameters based on RTCP feedback to maximize encoding quality. | ||
|
||
Encryption is mandatory in WebRTC, therefore secure end-to-end transport of media is implicit. | ||
|
||
This document proposes a simple HTTP based protocol that will allow WebRTC based ingest of content into streaming servics and/or CDNs. | ||
This document proposes a simple HTTP based protocol that will allow WebRTC based ingest of content into streaming services and/or CDNs. | ||
|
||
## Current Draft | ||
- [draft-ietf-wish-whip-00](https://datatracker.ietf.org/doc/html/draft-ietf-wish-whip-00) | ||
- [draft-ietf-wish-whip-00](https://datatracker.ietf.org/doc/html/draft-ietf-wish-whip-01) |
This file contains bidirectional Unicode text that may be interpreted or compiled differently than what appears below. To review, open the file in an editor that reveals hidden Unicode characters.
Learn more about bidirectional Unicode characters