AVTCORE S. Garcia Murillo Internet-Draft A. Gouaillard Intended status: Standards Track CoSMo Software Expires: August 23, 2021 February 19, 2021 Codec agnostic RTP payload format for video draft-gouaillard-avtcore-codec-agn-rtp-payload-00 Abstract RTP Media Chains usually rely on piping encoder output directly to packetizers. Media packetization formats often support a specific codec format and optimize RTP packets generation accordingly. With the development of Selective Forward Unit (SFU) solutions, that do not process media content server side, the need for media content processing at the origin and at the destination has arised. RTP Media Chains used e.g. in WebRTC solutions are increasingly relying on application-specific transforms that sit in-between encoder and packetizer on one end and in-between depacketizer and decoder on the other end. This use case has become so important, that the W3C is standardizing the capacity to access encoded content with the [WebRTCInsertableStreams] API proposal. An extremely popular use case is application level end-to-end encryption of media content, using for instance [SFrame]. Whatever the modification applied to the media content, RTP packetizers can no longer expect to use packetization formats that mandate media content to be in a specific codec format. In the extreme cases like encryption, where the RTP Payload is made completely opaque to the SFUs, some extra mechanism must also be added for them to be able to route the packets without depending on RTP payload or payload headers. The traditionnal process of creating a new RTP Payload specification per content would not be practical as we would need to make a new one for each codec-transform pair. This document describes a solution, which provides the following features in the case the encoded content has been modified before reaching the packetizer: - a paylaod agnostic RTP packetization format that can be used on any media content, - a signalling mechanism for the above format and the inner payload, Both of the above mechanism are backward compatible with most of (S)RTP/RTCP mechanisms used for bandwidth estimation and congestion control in Garcia Murillo & GouaillaExpires August 23, 2021 [Page 1] Internet-Draft Codec agnostic RTP payload format for video February 2021 RTP/SRTP/webrtc, including but not limited to SSRC, RED, FEC, RTX, NACK, SR/RR, REMB, transport-wide-CC, TIMBR, .... It as illustrated by existing implementations in chrome, safari, and Medooze. This document also describes a solution to allow SFUs to continue performing packet routing on top of this generic RTP packetization format. This document complements the SFrame (media encryption), and Dependency Descriptor (AV1 payload annex) documents to provide an End-to-End-Encryption solution that would sit on top of SRTP/Webrtc, use SFUs on the media back-end, and leverage W3C APIs in the browser. A high level description of such system will be provided as an informational I-D in the SFrame WG and then cited here. Status of This Memo This Internet-Draft is submitted in full conformance with the provisions of BCP 78 and BCP 79. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF). Note that other groups may also distribute working documents as Internet-Drafts. The list of current Internet- Drafts is at https://datatracker.ietf.org/drafts/current/. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet-Drafts as reference material or to cite them other than as "work in progress." This Internet-Draft will expire on August 23, 2021. Copyright Notice Copyright (c) 2021 IETF Trust and the persons identified as the document authors. All rights reserved. This document is subject to BCP 78 and the IETF Trust's Legal Provisions Relating to IETF Documents (https://trustee.ietf.org/license-info) in effect on the date of publication of this document. Please review these documents carefully, as they describe your rights and restrictions with respect to this document. Code Components extracted from this document must include Simplified BSD License text as described in Section 4.e of the Trust Legal Provisions and are provided without warranty as described in the Simplified BSD License. Garcia Murillo & GouaillaExpires August 23, 2021 [Page 2] Internet-Draft Codec agnostic RTP payload format for video February 2021 Table of Contents 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . 3 2. Goals . . . . . . . . . . . . . . . . . . . . . . . . . . . . 6 3. RTP Packetization . . . . . . . . . . . . . . . . . . . . . . 6 4. Payload Multiplexing . . . . . . . . . . . . . . . . . . . . 7 5. SDP Negotiation . . . . . . . . . . . . . . . . . . . . . . . 8 6. SFU Packet Selection . . . . . . . . . . . . . . . . . . . . 9 7. Redundancy Techniques Considerations . . . . . . . . . . . . 10 7.1. Retransmission Techniques . . . . . . . . . . . . . . . . 10 7.2. Forward Error Correction (FEC) Techniques . . . . . . . . 10 7.3. Redundant Audio Data Techniques . . . . . . . . . . . . . 10 8. Alternatives . . . . . . . . . . . . . . . . . . . . . . . . 11 8.1. Generic Packetization With In-Payload APT . . . . . . . . 11 8.2. A Payload Type for Generic Packetization AND Media Format 11 8.3. A RTP Header To Choose Packetization . . . . . . . . . . 13 9. Security Considerations . . . . . . . . . . . . . . . . . . . 13 10. IANA Considerations . . . . . . . . . . . . . . . . . . . . . 14 10.1. Registration of audio/generic . . . . . . . . . . . . . 14 11. Registration of video/generic . . . . . . . . . . . . . . . . 14 12. References . . . . . . . . . . . . . . . . . . . . . . . . . 15 12.1. Normative References . . . . . . . . . . . . . . . . . . 15 12.2. Informative References . . . . . . . . . . . . . . . . . 15 Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . 16 1. Introduction As per Figure 1 of [RFC7656], a Media Packetizer transforms a single Encoded Stream into one or several RTP packets. The Encoded Stream is coming straight from the Media Encoder and is expected to follow the format produced by the Media Encoder. A number of Media Packetizer formats have been designed to process a specific format produced by Media Encoder. For instance [RFC6184] is dedicated to the processing of content produced by H.264 Media Encoders, and generates packets following NALUs organization. WebRTC applications are increasingly deploying end-to-end encryption solutions on top of RTP Media Chains. End-to-end encryption is implemented by inserting application-specific Media Transformers between Media Encoder and Media Packetizer on the sending side, and between Media Depacketizer and Media Decoder on the receiving side, as described in Figure 1 and Figure 2. To support end-to-end encryption, Media Transformers can use the [SFrame] format. In browsers, Media Transformers are implemented using [WebRTCInsertableStreams], for instance by injecting JavaScript code provided by web pages. Garcia Murillo & GouaillaExpires August 23, 2021 [Page 3] Internet-Draft Codec agnostic RTP payload format for video February 2021 Physical Stimulus | V +----------------------+ | Media Capture | +----------------------+ | Raw Stream V +----------------------+ | Media Source |<-- Synchronization Timing +----------------------+ | Source Stream V +----------------------+ | Media Encoder | +----------------------+ | Encoded Stream V +----------------------+ | Media Transformer |<-- NEW: application-specific transform +----------------------+ (e.g. SFrame Encryption) | Transformed Stream +------------+ V | V +----------------------+ | +----------------------+ | Media Packetizer | | | RTP-Based Redundancy | +----------------------+ | +----------------------+ | | | +-------------+ Redundancy RTP Stream Source RTP Stream | V V +----------------------+ +----------------------+ | RTP-Based Security | | RTP-Based Security | +----------------------+ +----------------------+ | | Secured RTP Stream Secured Redundancy RTP Stream V V +----------------------+ +----------------------+ | Media Transport | | Media Transport | +----------------------+ +----------------------+ Figure 1: Sender Side Concepts in the Media Chain With Application-level Media Transform Garcia Murillo & GouaillaExpires August 23, 2021 [Page 4] Internet-Draft Codec agnostic RTP payload format for video February 2021 These RTP packets are sent over the wire to a receiver media chain matching the sender side, reaching the Media Depacketizer that will reconstruct the Encoded Stream before passing it to the Media Decoder. +----------------------+ +----------------------+ | Media Transport | | Media Transport | +----------------------+ +----------------------+ Received | Received | Secured Secured RTP Stream Redundancy RTP Stream V V +----------------------+ +----------------------+ | RTP-Based Validation | | RTP-Based Validation | +----------------------+ +----------------------+ | | Received RTP Stream Received Redundancy RTP Stream | | | +--------------------+ V V +----------------------+ | RTP-Based Repair | +----------------------+ | Repaired RTP Stream V +----------------------+ | Media Depacketizer | +----------------------+ | Received Transformed Stream V +----------------------+ | Media Transformer |<-- NEW: application-specific transform +----------------------+ (e.g. SFrame Decryption) | Received Encoded Stream V +----------------------+ | Media Decoder | +----------------------+ | Received Source Stream V +----------------------+ | Media Sink |--> Synchronization Information +----------------------+ | Received Raw Stream Garcia Murillo & GouaillaExpires August 23, 2021 [Page 5] Internet-Draft Codec agnostic RTP payload format for video February 2021 V +----------------------+ | Media Render | +----------------------+ | V Physical Stimulus Figure 2: Receiver Side Concepts in the Media Chain With Application-level Media Transform This generic packetization does not change how the mapping between one or several encoded or dependant streams are mapped to the RTP streams or how the synchronization sources(s) (SSRC) are assigned. Given the use of post-encoder application-specific transforms, the whole Media Chain needs to be made aware of it. This includes the sender post-transform Media Chain, Media Transport intermediaries (SFUs typically) and receiver pre-transform Media Chain. As these transforms can alter Encoded Streams in any possible way, the use of codec-specific Media Packetizers like [RFC6184] on Transformed Stream may be suboptimal on sender side. It may also be problematic on the receiving side in case codec-specific processing is done prior the Media Transformer. Media Transport intermediaries are often looking at the Media Content itself to fuel their packet selection algorithms. 2. Goals The objective of this document is to support inserting any application-specific transform between encoders and packetizers in the Media Chain. For that purpose, this document will: 1. Provide a generic packetization format that supports any media content (compressed audio, compressed video, encrypted content...) that allows reuse of existing RTP mechanisms in place in WebRTC applications such as RTX, RED or FEC. 2. Provide a way to negotiate use of the generic packetization format between sender and receiver, with minimum impact on existing negotiation approaches. 3. Provide a side-channel information so that network intermediaries (SFU in particular) can do their existing packet routing strategies without inspecting the media content. 3. RTP Packetization A generic packetizer, by design, is not expected to understand the format of the media to transmit. The unit used by the packetizer to do processing is called a frame in the remainder of the document. Garcia Murillo & GouaillaExpires August 23, 2021 [Page 6] Internet-Draft Codec agnostic RTP payload format for video February 2021 It is the responsibility of the application using the packetizer to group media content in meaningful frames. In the common case of a video codec, the packetizer frame is the frame in byte format (h264 annex b for example) generated by the encoder. If the application wants to transform encoded content, the application needs to split the encoded content into frames prior the transform. Each frame is then transformed independently, for instance encrypted using [SFrame]. The content of each transformed frame is then processed by the packetizer. In the case of a video codec supporting spatial scalability, each spatial layer MUST be split in its own frame by the application before passing it to the packetizer. When the packetizer receives a frame from the application, it MUST fragment the frame content in multiple RTP packets to ensure packets do not exceed the network maximum transmission unit. The content of the frame will be treated as a binary blob by the packetizer, so the decision about the boundaries of each fragment is decided arbitrarily by the packetizer. The packetizer or any relying server MUST NOT modify the frame content and concatenating the RTP payload of the RTP packets for each frame MUST produce the exact binary content of the input frame content. The marker bit of each RTP packet in a frame MUST be set according to the audio and video profiles specified in [RFC3551]. The spatial layer frames are sent in ascending order, with the same RTP timestamp, and only the last RTP packet of the last spatial layer frame will have the marker bit set to 1. 4. Payload Multiplexing In order to reduce the number of payload type in the SDP exchange, a single payload type code for the generic packetization can be used for all negotiated media formats. That requires to identify the original payload type code of the frame negotiated media format, called the associated payload type (APT) hereunder. The APT value is the payload type code of the associated format passed to the generic Media Packetizer before any transformation is applied. The APT value is sent in a dedicated header extension. The payload of this header extension can be encoded using either the one-byte or two-byte header defined in [RFC5285]. Figures 3 and 4 show examples with each one of these examples. Garcia Murillo & GouaillaExpires August 23, 2021 [Page 7] Internet-Draft Codec agnostic RTP payload format for video February 2021 0 1 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | ID | len=0 |S| APT | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Figure 3: Frame Associated Payload Type Encoding Using the One-Byte Header Format 0 1 2 3 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ | ID | len=1 |S| APT | 0 (pad) | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ Figure 4: Frame Associated Payload Type Encoding Using the Two-Byte Header Format The APT value is the associated payload type value. The S bit indicates if the media stream can be forwarded safely starting from this RTP packet. Typically, it will be set to 1 on the first RTP packet of an intra video frame and in all RTP audio packets. Receivers MUST be ready to receive RTP packets with different associated payload types in the same way they would receive different payload type codes on the RTP packets. The URI for declaring this header extension in an extmap attribute is "urn:ietf:params:rtp-hdrext:associated-payload-type". 5. SDP Negotiation To use the RTP generic packetization, the SDP Offer/Answer exchange MUST negotiate: - The payload type of the negotiated codec format - The generic payload type - The associated payload type header extension Only the negotiated payload types are allowed to be used as associated payload types. Figure 5 illustrates a SDP that negotiates exchange of video using either VP8 or VP9 codecs with the possibility to use the generic packetization. In this example, RTX is also negotiated and will be applied normally on each associated payload type. Garcia Murillo & GouaillaExpires August 23, 2021 [Page 8] Internet-Draft Codec agnostic RTP payload format for video February 2021 m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 a=setup:actpass a=mid:1 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=extmap:4 urn:ietf:params:rtp-hdrext:associated-payload-type a=sendrecv a=rtpmap:96 vp9/90000 a=rtpmap:97 vp8/90000 a=rtpmap:98 generic/90000 a=rtpmap:99 rtx/90000 a=fmtp:99 apt=96 a=rtpmap:100 rtx/90000 a=fmtp:100 apt=97 a=rtpmap:101 rtx/90000 a=fmtp:101 apt=98 Figure 5: SDP example negotiating the generic payload type and related header extension for video 6. SFU Packet Selection SFUs need to have a basic understanding of each frame they receive so they can decide to forward it or not and to which endpoint. They might need similar information to support media content recording. This information is either generic to a group of frame (called a stream hereafter) or specific to each frame. The information is transmitted as a RTP header extension as the RTP packet payload should be treated as opaque by the SFU. This is especially necessary if the payload is end-to-end encrypted. The amount of information should be limited to what is strictly necessary to the SFU task since it is not always as trusted as individual peers. For audio, configuration information such as Opus TOC might be useful. For video, configuration information might include: - Stream configuration information: resolution, quality, frame rate... - Codec specific configuration information: codec profile like profile_idc... - Frame specific information: whether the stream is decodable when starting from this frame, whether the frame is skippable... For video content, this information can be sent using a Dependency Descriptor header extension. In that case, the first RTP packet of Garcia Murillo & GouaillaExpires August 23, 2021 [Page 9] Internet-Draft Codec agnostic RTP payload format for video February 2021 the frame will have its start_of_frame equal to 1 and the last packet will have its end_of_frame equal to 1. 7. Redundancy Techniques Considerations The solution described in this document is expected to integrate well with the existing RTP ecosystem. This section describes how the generic packetizer can be used jointly with existing techniques that allow to mitigate unreliable transports. 7.1. Retransmission Techniques [RFC4588] defines a retransmission payload format (RTX) that can be used in case of packet loss. As defined in [RFC4588], RTX is able to handle any payload format, including the format described in this document. Given RTX preserves both RTP packet payload and headers, the receiver will be able to identify the payload type of the recovered packet and whether generic packetization is used. RTX will also allow recovering RTP header extensions that convey information on the media content itself. 7.2. Forward Error Correction (FEC) Techniques FEC is another technique used in RTP Media Chains to protect media content against packet loss. [RFC5109] defines such a payload format used to transmit FEC for specific packets protection. FEC may protect some parts of the media content more than others. For instance, intra video frame encoded data or important network abstraction layer units (NALUs) like SPS/PPS may be more protected. With a post-encoder transform and the use of a generic packetization, the granularity of the recovery mechanism is no longer at the NALU level but at the level of the frame generated by the post-encoder transform. In case a SVC codec is used, each spatial layer will be processed as an independent frame. In that case, base layers can be protected more heavily than higher resolution layers. 7.3. Redundant Audio Data Techniques As defined in [RFC7656] RTP-based redundancy is defined here as a transformation that generates redundant or repair packets sent out as a Redundancy RTP Stream to mitigate Network Transport impairments, like packet loss and delay. [RFC2198] defines a payload format for sending the same audio data encoded multiple times at different quality levels. This allows to use a lower quality encoding of the audio data, should the higher quality encoding of the audio data is lost during the transmission. Garcia Murillo & GouaillaExpires August 23, 2021 [Page 10] Internet-Draft Codec agnostic RTP payload format for video February 2021 If a Media Transformation is in use, both the primary and redundant encoding must be transformed independently and the redundant packet created normally. As the RTP headers present in the redundant packet are only applicable to the primary encoding, if the payload type for a redundant encoding block is mapped to the generic packetizer, the value of the associated payload type for the primary encoding is applied to the redundant encoding block as well. 8. Alternatives Various alternatives can be used to implement and negotiate generic packetization. This section describes a few additional alternatives. This section is to be removed before finalization of the document. 8.1. Generic Packetization With In-Payload APT Instead of using a RTP header extension to convey the APT value, it is prepended in the RTP payload itself. As the value cannot change for a whole frame, its value is prepended to the first packet generated of the frame only. This removes the need to negotiate a dedicated header extension, but may require the SFU to update the payload when sending or recording content. 8.2. A Payload Type for Generic Packetization AND Media Format The payload type is negotiated in the SDP so as to identify both the negotiated codec format and the generic packetization use. There is no network cost but this increases the number of payload types used in the SDP. Garcia Murillo & GouaillaExpires August 23, 2021 [Page 11] Internet-Draft Codec agnostic RTP payload format for video February 2021 m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 a=setup:actpass a=mid:1 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=rtpmap:96 vp9/90000 a=rtpmap:97 generic/90000 a=fmtp:97 apt=96 a=rtpmap:98 vp8/90000 a=rtpmap:99 generic/90000 a=fmtp:99 apt=98 a=rtpmap:100 rtx/90000 a=fmtp:100 apt=96 a=rtpmap:101 rtx/90000 a=fmtp:101 apt=97 a=rtpmap:102 rtx/90000 a=fmtp:102 apt=98 a=rtpmap:103 rtx/90000 a=fmtp:103 apt=99 Figure 6: SDP example negotiating a payload type for format and generic packetization A variation of this approach is to consider defining generic payload types, each of them having an identified codec format. m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 a=setup:actpass a=mid:1 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=sendrecv a=rtpmap:96 generic/90000 a=fmtp:96 codec=vp9 a=rtpmap:97 generic/90000 a=fmtp:97 codec=vp8 a=rtpmap:98 rtx/90000 a=fmtp:98 apt=96 a=rtpmap:99 rtx/90000 a=fmtp:99 apt=97 Garcia Murillo & GouaillaExpires August 23, 2021 [Page 12] Internet-Draft Codec agnostic RTP payload format for video February 2021 Figure 7: SDP example negotiating a payload type for format and generic packetization 8.3. A RTP Header To Choose Packetization A RTP header extension can be used to flag content as opaque so that the receiver knows whether to use or not the generic packetization. As for the API header extension, the RTP header extension may not need to be sent for every packet, it could for instance be sent for the first packet of every intra video frame. The main advantage of this approach is the reduced impact on SDP negotiation. m=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 c=IN IP4 0.0.0.0 a=rtcp:9 IN IP4 0.0.0.0 a=setup:actpass a=mid:1 a=extmap:1 urn:ietf:params:rtp-hdrext:sdes:mid a=extmap:2 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id a=extmap:4 urn:ietf:params:rtp-hdrext:generic-packetization-use a=sendrecv a=rtpmap:96 vp9/90000 a=rtpmap:97 vp8/90000 a=rtpmap:98 rtx/90000 a=fmtp:98 apt=96 a=rtpmap:99 rtx/90000 a=fmtp:99 apt=97 Figure 8: SDP example negotiating generic packetization as RTP header extension 9. Security Considerations RTP packets using the payload format defined in this specification are subject to the general security considerations discussed in [RFC3550]. It is not expected that the proposed solutions (generic packetization and header extension) presented in this document can create new security threats. The use and implementation of RTP Media Chains containing Media Transformers needs to be done carerefully. It is important to refer to the security considerations discussed in [SFrame] and [WebRTCInsertableStreams]. In particular Media Transformers on the receiver side need to be prepared to receive arbitrary content, like decoders already do. Similarly, since Media Transformers can be implemented as JavaScript in browsers, RTP Packetizers should be prepared to receive arbitrary content. Garcia Murillo & GouaillaExpires August 23, 2021 [Page 13] Internet-Draft Codec agnostic RTP payload format for video February 2021 10. IANA Considerations Two new media subtypes have been registered with IANA, as described in this section. 10.1. Registration of audio/generic Type name: audio Subtype name: generic Required parameters: none Optional parameters: none Encoding considerations: This format is framed (see Section 4.8 in the template document) and contains binary data. Security considerations: TBD. Interoperability considerations: TBD Published specification: TBD. Applications that use this media type: TBD. Additional information: none Intended usage: COMMON Restrictions on usage: TBD Author: Change controller: 11. Registration of video/generic Type name: video Subtype name: generic Required parameters: none Optional parameters: none Encoding considerations: This format is framed (see Section 4.8 in the template document) and contains binary data. Garcia Murillo & GouaillaExpires August 23, 2021 [Page 14] Internet-Draft Codec agnostic RTP payload format for video February 2021 Security considerations: TBD. Interoperability considerations: TBD Published specification: TBD. Applications that use this media type: TBD. Additional information: none Intended usage: COMMON Restrictions on usage: TBD Author: Change controller: 12. References 12.1. Normative References [RFC3550] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson, "RTP: A Transport Protocol for Real-Time Applications", STD 64, RFC 3550, DOI 10.17487/RFC3550, July 2003, . [RFC3551] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video Conferences with Minimal Control", STD 65, RFC 3551, DOI 10.17487/RFC3551, July 2003, . [RFC5285] Singer, D. and H. Desineni, "A General Mechanism for RTP Header Extensions", RFC 5285, DOI 10.17487/RFC5285, July 2008, . [RFC7656] Lennox, J., Gross, K., Nandakumar, S., Salgueiro, G., and B. Burman, Ed., "A Taxonomy of Semantics and Mechanisms for Real-Time Transport Protocol (RTP) Sources", RFC 7656, DOI 10.17487/RFC7656, November 2015, . 12.2. Informative References Garcia Murillo & GouaillaExpires August 23, 2021 [Page 15] Internet-Draft Codec agnostic RTP payload format for video February 2021 [RFC2198] Perkins, C., Kouvelas, I., Hodson, O., Hardman, V., Handley, M., Bolot, J., Vega-Garcia, A., and S. Fosse- Parisis, "RTP Payload for Redundant Audio Data", RFC 2198, DOI 10.17487/RFC2198, September 1997, . [RFC4588] Rey, J., Leon, D., Miyazaki, A., Varsa, V., and R. Hakenberg, "RTP Retransmission Payload Format", RFC 4588, DOI 10.17487/RFC4588, July 2006, . [RFC5109] Li, A., Ed., "RTP Payload Format for Generic Forward Error Correction", RFC 5109, DOI 10.17487/RFC5109, December 2007, . [RFC6184] Wang, Y., Even, R., Kristensen, T., and R. Jesup, "RTP Payload Format for H.264 Video", RFC 6184, DOI 10.17487/RFC6184, May 2011, . [SFrame] "Secure Frame (SFrame)", n.d., . [WebRTCInsertableStreams] "WebRTC Insertable Media using Streams", n.d., . Authors' Addresses Sergio Garcia Murillo CoSMo Software Email: sergio.garcia.murillo@cosmosoftware.io Alexandre Gouaillard CoSMo Software Email: alex.gouaillard@cosmosoftware.io Garcia Murillo & GouaillaExpires August 23, 2021 [Page 16]