Code change:
The text says: "The media codecs used in older protocols do not always match those being used in WebRTC, mandating transcoding on the ingest node," I don't understand this sentence. If I use an "older protocol", e.g, RTMP, why do I need to care about WebRTC codecs?
That refers if you use RTMP for ingest and WebRTC for viewing.
rewriten the editorial in
The text says: "These protocols are much older than webrtc and lack by default some important security and resilience features provided by webrtc with minimal delay." I assume you refer to RTMP, which is mentioned in the previous Section. Doesn't RTMPS solve some of the security issues? But in general, if you want to put up WHIP against RTMP, I think you need a little more text. Also, as WHIP is supposed to be an "easy to implement" protocol, I think it would be fair to also list some of the functions NOT supported compared to e.g., RTMP. As I have mentioned before, there is no command for start the ingestion - it is assumed to have begun when the connection has been established. Also, there is no PAUSE/RESUME.
It is more referring to RTSP which lacks DTLS support. no command for start the ingestion - it is assumed to have begun when the connection has been established. Also, there is no PAUSE/RESUME. RTMP protocol was intended as an RPC mechanism between the flash player and server. It allowed to create custom apps both on client and server side, which supported a wide range of methods using AMF. However, current "RTMP protocol" implemented for ingesting is a reversed engineered version of the implementation done by the FMS and it lacks support for PAUSE/RESUME and the publication is started from the start of the RTMP connection (same as whip).
rewriten the editorial in

Related discussion:
text: based on RTP and may be the closest in terms of features to WebRTC, is not compatible with the SDP offer/answer model RFC3264. <del> In the specific case of media ingestion into a streaming service, some assumptions can be made about the server-side which simplifies the WebRTC compliance burden, as detailed in WebRTC-gateway document draft-ietf-rtcweb-gateways. </del> <ins> So, currently, there is no standard protocol designed for ingesting media into a streaming service using WebRTC and so content providers still rely heavily on protocols like RTMP for doing so.  Most of those protocols are not RTP based, requiring media protocol translation when doing egress via WebRTC.  Avoiding this media protocol translation is desirable as there is no functional parity between those protocols and WebRTC and it increases the implementation complexity at the media server side. Also, the media codecs used in those protocols tend to be limited and not negotiated, not always matching the mediac codes supported in WebRTC.  This requires to perform transcoding on the ingest node, which introduces delay, degrades media quality and increases the processing workload required on the server side.  Server side transcoding that has traditionally been done to present multiple renditions in Adaptive Bit Rate Streaming (ABR) implementations can be replaced with Simulcast RFC8853 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. </ins> This document proposes a simple protocol for supporting WebRTC as media ingestion method which: