2. RTCP Multiplexing – WebRTC supports multiplex of both audio/video and RTP/RTCP over the same RTP session and port, this is not supported in IMS so is necessary to perform the demultiplexing. The Real-time Transport Protocol (RTP) [] is generally used to carry real-time media for conversational media sessions, such as video conferences, across the Internet. WebRTC uses a variety of protocols, including Real-Time Transport Protocol (RTP) and Real-Time Control Protocol (RTCP). e. But now I am confused about which byte I should measure. 一、webrtc. To disable WebRTC in Firefox: Type about:config in the address bar and press Enter. WebRTC uses Opus and G. Kubernetes has been designed and optimized for the typical HTTP/TCP Web workload, which makes streaming workloads, and especially UDP/RTP based WebRTC media, feel like a foreign citizen. In twcc/send-side bwe the estimation happens in the entity that also encodes (and has more context) while the receiver is "simple". If works then you can add your firewall rules for WebRTC and UDP ports . Attempting to connect Freeswitch + WebRTC with RTMP and jssip utilizing NAT traversal via STUN servers . With this example we have pre-made GStreamer and ffmpeg pipelines, but you can use any tool you like! This approach allows for recovery of entire RTP packets, including the full RTP header. With it, you can configure the encoding used for the corresponding track, get information about the device's media capabilities, and so forth. OpenCV was designed for computational efficiency and with a strong focus on real-time applications. The way this is implemented in Google's WebRTC implementation right now is this one: Keep a copy of the packets sent in the last 1000 msecs (the "history"). You cannot use WebRTC to pick the RTP packets and send them over a protocol of your choice, like WebSockets. 1. Point 3 says, Media will use TCP or UDP, but DataChannel will use SCTP, so DataChannel should be reliable, because SCTP is reliable (according to the SCTP RFC ). Just as WHIP takes care of the ingestion process in a broadcasting infrastructure, WHEP takes care of distributing streams via WebRTC instead. WebRTC, Web Real-time communication is the protocol (collection of APIs) that allows direct communication between browsers. 1. Check the Try to decode RTP outside of conversations checkbox. The protocol is “built” on top of RTP as a secure transport protocol for real time media and is mandated for use by. t. Two commonly used real-time communication protocols for IP-based video and audio communications are the session initiation protocol (SIP) and web real-time communications (WebRTC). Edit: Your calculcations look good to me. You will need specific pipeline for your audio, of course. A connection is established through a discovery and negotiation process called signaling. In this guide, we'll examine how to add a data channel to a peer connection, which can then be used to securely exchange arbitrary data; that is, any kind of data we wish, in any format we choose. and for that WebSocket is a likely choice. WebRTC is an open-source project that enables real-time communication capabilities for web and mobile applications. Review. Alex Gouaillard and his team at CoSMo Software put together a load test suite to measure load vs. I'm studying WebRTC and try to figure how it works. VNC is used as a screen-sharing platform that allows users to control remote devices. The more simple and straight forward solution is use a media server to covert RTMP to WebRTC. This memo describes the media transport aspects of the WebRTC framework. Conclusion. WebRTC is a free, open project that enables web. WebRTC is a modern protocol supported by modern browsers. Because the WebRTC is not only RTP, but also need to transcode the audio from opus to aac, and there is something like the jitter-buffer, NACK or packet out-of-order to handle. A similar relationship would be the one between HTTP and the Fetch API. RTSP is commonly used for streaming media, such as video or audio streams, and is best for media that needs to be broadcasted in real-time. You signed in with another tab or window. The outbound is the stream from the server to the. SIP is a protocol, not an API; whereas WebRTC is an API, with an associated set of protocols. Although RTP is called a transport protocol, it’s an application-level protocol that runs on top of UDP, and theoretically, it can run on top of any other transport protocol. Click on settings. After the setup between the IP camera and server is completed, video and audio data can be transmitted using RTP. RTP is heavily used in latency critical environments like real time audio and video (its the media transport in SIP, H. The framework for Web Real-Time Communication (WebRTC) provides support for direct interactive rich communication using audio, video, text, collaboration, games, etc. RTP header vs RTP payload. While that’s all we need to stream, there are a few settings that you should put in for proper conversion from RTMP to WebRTC. With WebRTC, you can add real-time communication capabilities to your application that works on top of an open standard. RTP/RTSP, WebRTC HLS/DASH CMAF with LLC Streaming latency continuum 60+ seconds 45 seconds 30 seconds 18 seconds 05 seconds 02 seconds 500 ms. If you are connecting your devices to a media server (be it an SFU for group calling or any other. The WebRTC API makes it possible to construct websites and apps that let users communicate in real time, using audio and/or video as well as optional data and other information. You can think of Web Real-Time Communications (WebRTC) as the jack-of-all-trades up. A similar relationship would be the one between HTTP and the Fetch API. After loading the plugin and starting a call on, for example, appear. You should also forward the Sender Reports if you want to synchronize. Sorted by: 14. This is the main WebRTC pro. WebRTC is a Javascript API (there is also a library implementing that API). Key exchange MUST be done using DTLS-SRTP, as described in [RFC8827]. Note: RTSPtoWeb is an improved service that provides the same functionality, an improved API, and supports even more protocols. (which was our experience in converting FTL->RTMP). RTSP stands for Real-Time Streaming. In firefox, you can just call . Two commonly used real-time communication protocols for IP-based video and audio communications are the session initiation protocol (SIP) and web real-time communications (WebRTC). Note that STUNner itself is a TURN server but, being deployed into the same Kubernetes cluster as the game. 2 RTP R TP is the Internet-standard protocol for the transport of real-time data, including audio and video [6, 7]. It lists a. The legacy getStats() WebRTC API will be removed in Chrome 117, therefore apps using it will need to migrate to the standard API. Rate control should be CBR with a bitrate of 4,000. sdp latency=0 ! autovideosink This pipeline provides latency parameter and though in reality is not zero but small latency and the stream is very stable. Here’s how WebRTC compares to traditional communication protocols on various fronts: Protocol Overheads and Performance: Traditional protocols such as SIP and RTP are laden with protocol overheads that can affect performance. 1. WebRTC: To publish live stream by H5 web page. X. Other key management schemes MAY be supported. For example for a video conference or a remote laboratory. From a protocol perspective, in the current proposal the two protocols are very similar, and in fact. Wowza enables single port for WebRTC over TCP; Unreal Media Server enables single port for WebRTC over TCP and for WebRTC over UDP as well. enabled and double-click the preference to set its value to false. UDP vs TCP from the SIP POV TCP High Availability, active-passive Proxy: – move the IP address via VRRP from active to passive (it becomes the new active) – Client find the “tube” is broken – Client re-REGISTER and re-INVITE(replaces) – Location and dialogs are recreated in server – RTP connections are recreated by RTPengine from. A PeerConnection accepts a plugable transport module, so it could be an RTCDtlsTransport defined in webrtc-pc or a DatagramTransport defined in WebTransport. This pairing of send and. It supports sending data both unreliably via its datagram APIs, and reliably via its streams APIs. (RTP). The WebRTC API then allows developers to use the WebRTC protocol. This description is partially approximate, since VoIP in itself is a concept (and not a technological layer, per se): transmission of voices (V) over (o) Internet protocols (IP). Difficult to scale. Browser is installed on every workstation, so to launch a WebRTC phone, you just need to open the link and log in. It is HTML5 compatible and you can use it to add real-time media communications directly between browser and devices. unread, Apr 29, 2013, 1:26:59 PM 4/29/13. And I want to add some feature, like when I. A. 实时音视频通讯只靠UDP. It proposes a baseline set of RTP. ). between two peers' web browsers. . With WebRTC, developers can create applications that support video, audio, and data communication through a set of APIs. In contrast, WebRTC is designed to minimize overhead, with a more efficient and streamlined communication. One significant difference between the two protocols lies in the level of control they each offer. My goal now is to take this audio-stream and provide it (one-to-many) to different Web-Clients. In such cases, an application level implementation of SCTP will usually be used. First thing would be to have access to the media session setup protocol (e. Both mediasoup-client and libmediasoupclient need separate WebRTC transports for sending and receiving. Because as far as I know it is not designed for. The advantage of RTSP over SIP is that it's a lot simpler to use and implement. You can get around this issue by setting the rtcpMuxPolicy flag on your RTCPeerConnections in Chrome to be “negotiate” instead of “require”. For interactive live streaming solutions ranging from video conferencing to online betting and bidding, Web Real-Time Communication (WebRTC) has become an essential underlying technology. Điều này cho phép các trình duyệt web không chỉ. Add a comment. Apparently so is HEVC. Most streaming devices that are ONVIF compliant allow RTP/RTSP streams to be initiated both within and separately from the ONVIF protocol. For something bidirectional, you should just pick WebRTC - its codecs are better, its availability is better. 3. On the other hand, WebRTC offers faster streaming experience with near real-time latency, and with its native support by. SRTP is simply RTP with “secure” in front: secure real-time protocol. Now perform the steps in Capturing RTP streams section but skip the Decode As steps (2-4). Thus we can say that video tag supports RTP(SRTP) indirectly via WebRTC. These two protocols have been widely used in softphone and video. Rate control should be CBR with a bitrate of 4,000. e. It describes a system designed to evaluate times at live streaming: establishment time and stream reception time from a single source to a large quantity of receivers with the use of smartphones. AFAIK, currently you can use websockets for webrtc signaling but not for sending mediastream. 3. RTP (=Real-Time Transport Protocol) is used as the baseline. 2. Click Restart when prompted. – Marc B. While WebSocket works only over TCP, WebRTC is primarily used over UDP (although it can work over TCP as well). Video conferencing and other interactive applications often use it. 265 and ISO/IEC International Standard 23008-2, both also known as High Efficiency Video Coding (HEVC) and developed by the Joint Collaborative Team on Video Coding (JCT-VC). 168. WebRTC requires some mechanism for finding peers and initiating calls. Decapsulate T140blocks from RTP packets sent by the SIP participant, and relay them (with or without translation to a different format) via data channels towards the WebRTC peer; Craft RTP packets to send to the SIP participant for every data sent via data channels by the WebRTC peer (possibly with translation to T140blocks);Pion is a WebRTC implementation written in Go and unlike Google’s WebRTC, Pion is specifically designed to be fast to build and customise. WebRTC is not supported and less reliable, less scalable compared to HLS. 3. With this example we have pre-made GStreamer and ffmpeg pipelines, but you can use any. The native webrtc stack, satellite view. This work presents a comparative study between two of the most used streaming protocols, RTSP and WebRTC. I've walkie-talkies sending the speech via RTP (G711a) into my LAN. Since RTP requires real-time delivery and is tolerant to packet losses, the default underlying transport protocol has been UDP, recently with DTLS on top to secure. One of the reasons why we’re having the conversation of WebRTC vs. TCP has complex state machinery to enable reliable bi-directional end-to-end packet flow assuming that intermediate routers and networks can have problems but. s. X. RTMP is good for one viewer. Redundant Encoding This approach, as described in [RFC2198], allows for redundant data to be piggybacked on an existing primary encoding, all in a single packet. August 10, 2020. Jul 15, 2015 at 15:02. t. While WebSocket works only over TCP, WebRTC is primarily used over UDP (although it can work over TCP as well). You can probably reduce some of the indirection, but I would use rtp-forwarder to take WebRTC -> RTP. With the growing demand for real-time and low-latency video delivery, SRT (secure and reliable transport) and WebRTC have become industry-leading technologies. RTCP protocol communicates or synchronizes metadata about the call. , SDP in SIP). For the review, we checked out both WHIP and WHEP on Cloudflare Stream: WebRTC-HTTP Ingress Protocol (WHIP) for sending a WebRTC stream INTO Cloudflare’s network as defined by IETF draft-ietf-wish-whip WebRTC-HTTP Egress Protocol (WHEP) for receiving a WebRTC steam FROM Cloudflare’s network as defined. WebRTC: Designed to provide Web Browsers with an easy way to establish 'Real Time Communication' with other browsers. The RTSPtoWeb {RTC} server opens the RTSP. Trunk State. make sure to set the ext-sip-ip and ext-rtp-ip in vars. github. Upon analyzing tcpdump, RTP from freeswitch to abonent is not visible, although rtp to freeswitch is present. 1. Janus is a WebRTC Server developed by Meetecho conceived to be a general purpose one. This is tied together in over 50 RFCs. RTP, known as Real-time Transport Protocol, facilitates the transmission of audio and video data across IP networks. Rather, it’s the security layer added to RTP for encryption. Stars - the number of stars that a project has on GitHub. (RTP) and Real-Time Control Protocol (RTCP). Beyond that they're entirely different technologies. 711 as audio codec with no optimization in its browser stack . ability to filter candidates using configuration in rtp. Published: 22 Apr 2015. 2. The Real-Time Messaging Protocol (RTMP) is a mature streaming protocol originally designed for streaming to Adobe Flash players. Every RTP packet contains a sequence number indicating its order in the stream, and timestamp indicating when the frame should be played back. It relies on two pre-existing protocols: RTP and RTCP. *WebRTC: As I'm trying to give a bigger audience the possibility to interact with each other, WebRTC is not suitable. What’s more, WebRTC operates on UDP allowing it to establish connections without the need for a handshake between the client and server. 실시간 전송 프로토콜 ( Real-time Transport Protocol, RTP )은 IP 네트워크 상에서 오디오와 비디오를 전달하기 위한 통신 프로토콜 이다. One moment, it is the only way to get real time media towards a web browser. WebRTC capabilities are most often used over the open internet, the same connections you are using to browse the web. RTSP: Low latency, Will not work in any browser (broadcast or receive). v. Rather, RTSP servers often leverage the Real-Time Transport Protocol (RTP) in. My answer to it in 2015 was this: There are two places where QUIC fits in WebRTC: 1. It was designed to allow for real-time delivery of video. Now it is time to make the peers communicate with each other. To initialize this process, RTCPeerConnection has two tasks: Ascertain local media conditions, such as resolution and codec capabilities. This article explains how to migrate your code, and what to do if you need more time to make this change. WebRTC uses the streaming protocol RTP to transmit video over the Internet and other IP networks. WebTransport is a web API that uses the HTTP/3 protocol as a bidirectional transport. In any case to establish a webRTC session you will need a signaling protocol also . I. send () for every chunk with no (or minimal) delay. yaml and ffmpeg commands for streaming. One of the best parts, you can do that without the need. which can work P2P under certain circumstances. As implemented by web browsers, it provides a simple JavaScript API which allows you to easily add remote audio or video calling to your web page or web app. Sean starts with TURN since that is where he started, but then we review ion – a complete WebRTC conferencing system – and some others. RTP and RTCP The Real-time Transport Protocol (RTP) [RFC3550] is REQUIRED to be implemented as the media transport protocol for WebRTC. g. HLS that outlines their concepts, support, and use cases. 12), so the only way to publish stream by H5 is WebRTC. Instead just push using ffmpeg into your RTSP server. This is achieved by using other transport protocols such as HTTPS or secure WebSockets. 1. ESP-RTC is built around Espressif's ESP32-S3-Korvo-2 multimedia development. HLS: Works almost everywhere. One of the first things for media encoders to adopt WebRTC is to have an RTP media engine. This document defines a set of ECMAScript APIs in WebIDL to extend the WebRTC 1. The API is based on preliminary work done in the W3C ORTC Community Group. 2020 marks the point of WebRTC unbundling. So transmitter/encoder is in the main hub and receiver/decoders are in the remote sites. The illustration above shows our “priorities” in how we’d like a session to connect in a peer to peer scenario. SIP over WebSocket (RFC 7118) – using the WebSocket protocol to support SIP signaling. WebRTC’s offer/answer model fits very naturally onto the idea of a SIP signaling mechanism. rtp-to-webrtc demonstrates how to consume a RTP stream video UDP, and then send to a WebRTC client. There is no any exact science behind this as you can be never sure on the actual limits, however 1200 byte is a safe value for all kind of networks on the public internet (including something like a double VPN connection over PPPoE) and for RTP there is no much. RTP itself. 264 or MPEG-4 video. RTSP is suited for client-server applications, for example where one. RTP (=Real-Time Transport Protocol) is used as the baseline. Thus main reason of using WebRTC instead of Websocket is latency. In contrast, VoIP takes place over the company’s network. simple API. WebSocket offers a simpler implementation process, with client-side and server-side components, while WebRTC involves more complex implementation with the need for signaling and media servers. Because as far as I know it is not designed for. Then your SDP with the RTP setup would look more like: m=audio 17032. Click the Live Streams menu, and then click Add Live Stream. However, end-to-end WebRTC encryption is totally possible. For an even terser description, also see the W3C definitions. WebRTC API. But WebRTC encryption is mandatory because real-time communication requires that WebRTC connections are established a. Use another signalling solution for your WebRTC-enabled application, but add in a signalling gateway to translate between this and SIP. RTP sends video and audio data in small chunks. For this example, our Stream Name will be Wowza HQ2. Sign in to Wowza Video. The stack will send the packets immediately once received from the recorder device and compressed with the selected codec. in, open the dev tools (Tools -> Web Developer -> Toggle Tools). WHEP stands for “WebRTC-HTTP egress protocol”, and was conceived as a companion protocol to WHIP. For live streaming, the RTMP is the de-facto standard in live streaming industry, so if you covert WebRTC to RTMP, you got everything, like transcoding by FFmpeg. Select a video file from your computer by hitting browse. It'll usually work. XDN architecture is designed to take full advantage of the Real Time Transport Protocol (RTP), which is the underlying transport protocol supporting both WebRTC and RTSP as well as IP voice communications. Describes methods for tuning Wowza Streaming Engine for WebRTC optimal. WebRTC technology is a set of APIs that allow browsers to access devices, including the microphone and camera. SCTP . As a native application you. Although RTP is called a transport protocol, it’s an application-level protocol that runs on top of UDP, and theoretically, it can run on top of any other transport protocol. The RTSPtoWeb add-on is a packaging of the existing project GitHub - deepch/RTSPtoWeb: RTSP Stream to WebBrowser which is an improved version of GitHub - deepch/RTSPtoWebRTC: RTSP. 1. Jakub has implemented an RTP Header extension making it possible to send colorspace information per frame; this enables. WebRTC stands for web real-time communications. In order to contact another peer on the web, you need to first know its IP address. During the early days of WebRTC there have been ongoing discussions if the mandatory video codec in. SSRC: Synchronization source identifier (32 bits) distinctively distinguishes the source of a data stream. FaceTime finally faces WebRTC – implementation deep dive. Until then it might be interesting to turn it off, it is enabled by default in WebRTC currently. ; In the search bar, type media. It thereby facilitates real-time control of the streaming media by communicating with the server — without actually transmitting the data itself. Similar to TCP, SCTP provides a flow control mechanism that makes sure the network doesn’t get congested SCTP is not implemented by all operating systems. Basically, it's like the square and rectangle concept; all squares are rectangles, but not all rectangles are. Add a comment. Mux Category: NORMAL The Mux Category is defined in [RFC8859]. October 27, 2022 by Traci Ruether When it comes to online video delivery, RTMP, HLS, MPEG-DASH, and WebRTC refer to the streaming protocols used to get content from. a video platform). We originally use the WebRTC stack implemented by Google and we’ve made it scalable to work on the server-side. RTSP is more suitable for streaming pre-recorded media. The Real-time Transport Protocol (RTP), defined in RFC 3550, is an IETF standard protocol to enable real-time connectivity for exchanging data that needs real-time priority. This signifies that many different layers of technology can be used when carrying out VoIP. WebRTC vs Mediasoup: What are the differences?. 5. WebRTC Latency. rtcp-mux is used by the vast majority of their WebRTC traffic. WebRTC connectivity. RTP is codec-agnostic, which means carrying a large number of codec types inside RTP is. WebRTC (Web Real-Time Communication) is a technology that enables Web applications and sites to capture and optionally stream audio and/or video media, as well as to exchange arbitrary data between browsers without requiring an intermediary. How does it work? # WebRTC uses two preexisting protocols RTP and RTCP, both defined in RFC 1889. It then uses the Real-Time Transport Protocol (RTP) in conjunction with Real-time Control Protocol (RTCP) for actually delivering the media stream. STUNner aims to change this state-of-the-art, by exposing a single public STUN/TURN server port for ingesting all media traffic into a Kubernetes. CSRC: Contributing source IDs (32 bits each) summate contributing sources to a stream which has been generated from multiple sources. Disable WebRTC on your browser . 711 which is common). Therefore to get RTP stream on your Chrome, Firefox or another HTML5 browser, you need a WebRTC server which will deliver the SRTP stream to browser. WebRTC to RTMP is used for H5 publisher for live streaming. That is why many of the solutions create a kind of end-to-end solution of a GW and the WebRTC. between two peers' web browsers. It sounds like WebSockets. Another special thing is that WebRTC doesn't specify the signaling. As such, it doesn't provide any functionality per se other than implementing the means to set up a WebRTC media communication with a browser, exchanging JSON messages with it, and relaying RTP/RTCP and messages between. The protocol is designed to handle all of this. By default, Wowza Streaming Engine transmuxes the stream into the HLS, MPEG-DASH, RTSP/RTP, and RTMP protocols for playback at scale. There's the first problem already. Then we jumped in to prepare an SFU and the tests. 应用层协议：RTP and RTCP. WebRTC takes the cake at sub-500 milliseconds while RTMP is around five seconds (it competes more directly with protocols like Secure Reliable Transport (SRT) and Real-Time Streaming Protocol. js and C/C++. I assume one packet of RTP data contains multiple media samples. In this article, we’ll discuss everything you need to know about STUN and TURN. WebRTC softphone runs in a browser, so it does not need to be installed separately. It is free streaming software. Creating Transports. Here’s how WebRTC compares to traditional communication protocols on various fronts: Protocol Overheads and Performance: Traditional protocols such as SIP and RTP are laden with protocol overheads that can affect performance. WebRTC vs. WebRTC does not include SIP so there is no way for you to directly connect a SIP client to a WebRTC server or vice-versa. This tutorial will guide you through building a two-way video-call. This is why Red5 Pro integrated our solution with WebRTC. In RFC 3550, the base RTP RFC, there is no reference to channel. Although the Web API is undoubtedly interesting for application developers, it is not the focus of this article. This guide reviews the codecs that browsers. As the speediest technology available, WebRTC delivers near-instantaneous voice and video streaming to and from any major browser. 4. You’ll need the audio to be set at 48 kilohertz and the video at a resolution you plan to stream at. Details regarding the video and audio tracks, the codecs. Let me tell you what we’ve done on the Ant Media Server side. This article describes how the various WebRTC-related protocols interact with one another in order to create a connection and transfer data and/or media among peers. T. However, the open-source nature of the technology may have the. These are protocols that can be used at contribution and delivery. 1. However, RTP does not. Current options for securing WebRTC include Secure Real-time Transport Protocol (SRTP) - Transport-level protocol that provides encryption, message authentication and integrity, and replay attack protection to the RTP data in both unicast and multicast applications. For testing purposes, Chrome Canary and Chrome Developer both have a flag which allows you to turn off SRTP, for example: cd /Applications/Google Chrome Canary. WebRTC (Web Real-Time Communication) [1] là một tiêu chuẩn định nghĩa tập hợp các giao thức truyền thông và các giao diện lập trình ứng dụng cho phép truyền tải thời gian thực trên các kết nối peer-to-peer. 3. 1. It is fairly old, RFC 2198 was written. 6. RTSP Stream to WebBrowser over WebRTC based on Pion (full native! not using ffmpeg or gstreamer). Adds protection, integrity, and message. One approach to ultra low latency streaming is to combine browser technologies such as MSE (Media Source Extensions) and WebSockets. WHEP stands for “WebRTC-HTTP egress protocol”, and was conceived as a companion protocol to WHIP. It offers the ability to send and receive voice and video data in real time over the network, usually no top of UDP. For example, to allow user to record a clip of camera to feedback for your product. In any case to establish a webRTC session you will need a signaling protocol also . The webrtc integration is responsible for signaling, passing the offer and an RTSP URL to the RTSPtoWebRTC server. But there’s good news. Vorbis is an open format from the Xiph. A forthcoming standard mandates that “require” behavior is used. Protocols are just one specific part of an. TWCC (Transport Wide Congestion Control) is a RTP extention of WebRTC protocol that is used for adaptive bitrate video streaming while mainteining a low transmission latency. conf to stop candidates from being offered and configuration in rtp. The WebRTC API is specified only for JavaScript. These APIs support exchanging files, information, or any data. When you get familiar with process above there are a couple of shortcuts you can apply in order to be more effective. The design related to codec is mainly in the Codec and RTP (segmentation / fragmentation) section. Whether it’s solving technical issues or regular maintenance, VNC is an excellent tool for IT experts. Billions of users can interact now that WebRTC makes live video chat easier than ever on the Web. WebRTC is HTML5 compatible and you can use it to add real-time media communications directly between browser and devices. Intermediary: WebRTC+WHIP with VP9 mode 2 (10bits 4:2:0 HDR) An interesting intermediate step if your hardware supports VP9 encoding (INTEL, Qualcomm and Samsung do for example).