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CHANGES.md

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# Changes
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## 14.0.0
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- fix: publish codec listings. (Todd Anderson)
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- fix: viewless subscriber for WHEPClient. (Todd Anderson)
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- feat: codec init param support for WHIP and RTC publishers. (Todd Anderson)
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## 13.0.0
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- fix: removal of SharedObject functionality. (Todd Anderson)
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- Fix for connectionParams tacked on WHIP/WHEP calls. (Todd Anderson)
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- Fix for Firefox issue with Bundle order (Todd Anderson)
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> The issue was that the offer provided by the server has a BUNDLE order of 0 1
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> 2. When the WHEP client set that as their remote SDP and generated an offer,
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> the BUNDLE order changed to 2 1 0 in the answer SDP. This would cause
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> subscription issues and non-existant candidates.
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> The issue was that the offer provided by the server has a BUNDLE order of 0 1 2. When the WHEP client set that as their remote SDP and generated an offer,
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> the BUNDLE order changed to 2 1 0 in the answer SDP. This would cause
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> subscription issues and non-existant candidates.
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> Solution was the munge in the incoming offer BUNDLE line order to that of the
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> answer.
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> Solution was the munge in the incoming offer BUNDLE line order to that of the
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> answer.
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- Using host query param when Location provided to WHIP/WHEP clients. (Todd Anderson)
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- Fix for live seek fullURL switching. (Todd Anderson)
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- using Hls.js fallback for Safari and seek feature. (bustardcelly)
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- Allow playback on scrub when unpublish (bustardcelly)
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> If the live stream becomes unpublished, but the video has been scrubbed to VOD time segment, then we want to allow for playback until the end of the live stream termination.
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> If the live stream becomes unpublished, but the video has been scrubbed to VOD time segment, then we want to allow for playback until the end of the live stream termination.
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## 10.6.0
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- Fix for flash stalled playback. (bustardcelly)
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- Added notifications for RTCPeerConnection open event. (bustardcelly)
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> This will allow for clients to recognize when they can start using peer connection APIs such as mute/unmute.
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> This will allow for clients to recognize when they can start using peer connection APIs such as mute/unmute.
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- fix for HLS socket authentication failure capture. (bustardcelly)
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- Fix for Safari Publisher issue. (bustardcelly)
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> When `gUM` returns with `video` track listed before `audio`, there is an issue on delivering the media to the server. Most likely due to mismatched candidates.
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> When `gUM` returns with `video` track listed before `audio`, there is an issue on delivering the media to the server. Most likely due to mismatched candidates.
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- Fixed improper spelling of `productInstallUrl` config option. (bustardcelly)
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> Is now `productInstallURL` for both rtmp publisher and subscriber.
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> Is now `productInstallURL` for both rtmp publisher and subscriber.
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- fix for unpublish not resolving if not currently publishing. (bustardcelly)
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- API update to allow for providing MediaStream for publish to bypass gUM in SDK internally. (bustardcelly)
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- fix for bandwidth inject in latest chrome browser. (bustardcelly)
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- adding notification and retry support for autoplay restriction of subscribers. (bustardcelly)
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> * WebRTC, RTMP and HLS Subscriber support
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> * muteOnAutoplayRestriction configuration property added
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> * Additional subscriber events for client-side notifications of autoplay restrictions
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> - WebRTC, RTMP and HLS Subscriber support
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> - muteOnAutoplayRestriction configuration property added
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> - Additional subscriber events for client-side notifications of autoplay restrictions
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- allowing for decoupled socket for sharedobject connections. (bustardcelly)
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- fix for DOM cleanup on unsubscribe. (bustardcelly)
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- Adding `rtcConfiguration` support (bustardcelly)
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> * WebRTC Publisher
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> * WebRTC Subscriber
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> * defaults to `iceServers` if rtcConfiguration is undefined
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> * Provides a default rtcConfiguration
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> - WebRTC Publisher
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> - WebRTC Subscriber
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> - defaults to `iceServers` if rtcConfiguration is undefined
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> - Provides a default rtcConfiguration
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## 5.2.0
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- remove of duplicate event handler. (bustardcelly)
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- Adding check for empty candidate from server (bustardcelly)
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> * Server at times (depending on client) will send a `candidate` object as empty during the negotiation process.
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> * A new event type has been added to notify listeners of this empty candidate
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> - Server at times (depending on client) will send a `candidate` object as empty during the negotiation process.
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> - A new event type has been added to notify listeners of this empty candidate
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- Pub/Sub support for mobile device orientation recognition. (bustardcelly)
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- removing onbeforeunload hook to dismantle sockets. (bustardcelly)
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- Allowing for auth/validation on HLS Subscribers (bustardcelly)
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> Enabling auth/validation using connectionParam options on HLS Subscribers. This will internally attempt to make a connection on a WebSocket using the connection parameters. If the socket is opened, it is considered successful validation. If rejected, the client is shutdown.
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> Enabling auth/validation using connectionParam options on HLS Subscribers. This will internally attempt to make a connection on a WebSocket using the connection parameters. If the socket is opened, it is considered successful validation. If rejected, the client is shutdown.
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- fix for mutiple property updates on shared object integration. (bustardcelly)
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- offload translation logic for orientation to focus on dynamic layout updates. (bustardcelly)
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- removal of unnecessary bandwidth config prop on subscribers. (bustardcelly)
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- Exposing autoLayoutOrientation (bustardcelly)
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> Defaulted to true.
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> When set to false, it is up to the developer to assign transitions as related to orientation of broadcasts.
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> Defaulted to true.
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> When set to false, it is up to the developer to assign transitions as related to orientation of broadcasts.
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## 4.5.0
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## 3.4.3
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- fix for test of `srcObject` in video element. (bustardcelly)
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- fix for subscriber.stop event dispatch. (bustardcelly)
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- offloading subscriber start event to websocket notification. (bustardcelly)
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- Update for API change in browsers. (bustardcelly)
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Though latest adapter.js should handle, adding for legacy and new browsers.
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Though latest adapter.js should handle, adding for legacy and new browsers.
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## 3.4.0
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## 3.3.0
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- using std camera sizes for RPRO-3787 (bustardcelly)
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- proper removal of child in flash embed. (bustardcelly)
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- support for IE and remove element on flash fallback. (bustardcelly)
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CONFERENCE_PARTICIPANT_README.md

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The 1 video track is a video composite of the video streams of all the `RTCConferenceParticipant`s in the group conference. Because of this, integrating a `RTCConferenceParticipant` into your application will most likely require a Mixer Node.
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> Read more about the [Red5 Pro Mixer](https://www.red5pro.com/docs/special/mixer/using-mixer/#gatsby-focus-wrapper).
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> Read more about the [Red5 Pro Mixer](https://www.red5.net/docs/special/mixer/using-mixer/#gatsby-focus-wrapper).
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* [Requirements](#requirements)
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* [Configuration Parameters](#configuration-parameters)
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## Mixer Node
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Because the video delivered to each `RTCConferenceParticipant` is a composite of all participants in the conference group, you will most likely need to deploy a [Red5 Pro Mixer] (https://www.red5pro.com/docs/special/mixer/using-mixer/#gatsby-focus-wrapper) as part of your solution.
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Because the video delivered to each `RTCConferenceParticipant` is a composite of all participants in the conference group, you will most likely need to deploy a [Red5 Pro Mixer] (https://www.red5.net/docs/special/mixer/using-mixer/#gatsby-focus-wrapper) as part of your solution.
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## Group Provision
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isRestricted: false,
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parameters: {
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audiotracks: 3,
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audiotracks: 3,
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videotracks: 1
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},
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restrictions: [],

PUBLISHER_README.md

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# Requirements
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The **Red5 Pro WebRTC SDK** is intended to communicate with a [Red5 Pro Server](https://www.red5pro.com/), which allows for broadcasting and consuming live streams utilizing [WebRTC](https://developer.mozilla.org/en-US/docs/Web/Guide/API/WebRTC) and other protocols, including [RTMP](https://en.wikipedia.org/wiki/Real_Time_Messaging_Protocol) and [HLS](https://en.wikipedia.org/wiki/HTTP_Live_Streaming).
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The **Red5 Pro WebRTC SDK** is intended to communicate with a [Red5 Pro Server](https://www.red5.net/), which allows for broadcasting and consuming live streams utilizing [WebRTC](https://developer.mozilla.org/en-US/docs/Web/Guide/API/WebRTC) and other protocols, including [RTMP](https://en.wikipedia.org/wiki/Real_Time_Messaging_Protocol) and [HLS](https://en.wikipedia.org/wiki/HTTP_Live_Streaming).
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As such, you will need a distribution of the [Red5 Pro Server](https://www.red5pro.com/) running locally or accessible from the web, such as [Amazon Web Services](https://www.red5pro.com/docs/server/awsinstall/).
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As such, you will need a distribution of the [Red5 Pro Server](https://www.red5.net/) running locally or accessible from the web, such as [Amazon Web Services](https://www.red5.net/docs/server/awsinstall/).
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> **[Click here to start using the Red5 Pro Server today!](https://account.red5.net/login)**
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| onGetUserMedia | [-] | [see below](#using-mediaconstraints-and-ongetusermedia) | An override method for performing your own `getUserMedia` request. |
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| signalingSocketOnly | [-] | `true` | Flag to indicate whether the `WebSocket` should only be used for signaling while establishing a connection. Afterward, all data between client and server will be sent over an `RTCDataChannel`.
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| dataChannelConfiguration | [-] | `{name: "red5pro"}` | An object used in configuring a n `RTCDataChannel`. _Only used when `signalingSocketOnly` is defined as `true`_ |
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| forceVP8 | [-] | `false` | Flag to force VP8 as the encoder for the outgoing stream. |
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| forceVP8 | [-] | `false` | Flag to force VP8 as the encoder for the outgoing stream. _Marked for Deprecation._ |
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| videoEncoding | [-] | `undefined` | `PublishVideoEncoder` enum: `VP8` | `H264` | `H265` . _Replacement of `forceVP8`._ |
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| audioEncoding | [-] | `undefined` | `PublishAudioEncoder` enum. |
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| endpoint | [-] | `undefined` | The full URL of the endpoint to stream to. **This is primarily used in Stream Manager 2.0 integration for clients.** [Refer to the Stream Manager 2.0 Section](#stream-manager-20)
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## WebRTC Example

SUBSCRIBER_README.md

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# Requirements
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The **Red5 Pro WebRTC SDK** is intended to communicate with a [Red5 Pro Server](https://www.red5pro.com/), which allows for broadcasting and consuming live streams utilizing [WebRTC](https://developer.mozilla.org/en-US/docs/Web/Guide/API/WebRTC) and other protocols, [HLS](https://en.wikipedia.org/wiki/HTTP_Live_Streaming).
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The **Red5 Pro WebRTC SDK** is intended to communicate with a [Red5 Pro Server](https://www.red5.net/), which allows for broadcasting and consuming live streams utilizing [WebRTC](https://developer.mozilla.org/en-US/docs/Web/Guide/API/WebRTC) and other protocols, [HLS](https://en.wikipedia.org/wiki/HTTP_Live_Streaming).
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As such, you will need a distribution of the [Red5 Pro Server](https://www.red5pro.com/) running locally or accessible from the web, such as [Amazon Web Services](https://www.red5pro.com/docs/server/awsinstall/).
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As such, you will need a distribution of the [Red5 Pro Server](https://www.red5.net) running locally or accessible from the web, such as [Amazon Web Services](https://www.red5.net/docs/server/awsinstall/).
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> **[Click here to start using the Red5 Pro Server today!](https://account.red5.net/login)**
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Flag to use the default playback controls of the SDK to play, pause, live scrub, and additional actions. If setting this to `false`, you will need to create and manage your own controls to interact with the live and VOD content.
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> You can also provide your own custom controls and/or class declarations easily following this [guideline](https://www.red5pro.com/docs/development/playbackcontrols/overview/).
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> You can also provide your own custom controls and/or class declarations easily following this [guideline](https://www.red5.net/docs/development/playbackcontrols/overview/).
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#### Video Encoding Configuration
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It is entirely possible to playback streams in HLS using a 3rd-Party library (such as [VideoJS](https://videojs.com/)), but you will not be able to do so while utilizing the Red5 Pro WebRTC SDK.
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> For more information on how to playback HLS in browsers without native support, please refer to the *Using VideoJS for Playback* section of the [Migration Guide](https://www.red5pro.com/docs/streaming/migrationguide.html#migrating-from-350-to-400).
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> For more information on how to playback HLS in browsers without native support, please refer to the *Using VideoJS for Playback* section of the [Migration Guide](https://www.red5.net/docs/streaming/migrationguide.html#migrating-from-350-to-400).
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## Initialization
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## Autoplay Restrictions
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In an attempt to provide a more pleasing user experience and reduce data consumption on mobile devices, browsers are continuing to evolve their `autoplay` policies. While generally and attempt to keep websites (read: *ads*) from playing back unwanted and/or unsolicited video and audio, these policies also affect those sites in which the sole intent _is to_ playback video and/or audio - such as from a conference web application built utilizing [Red5 Pro](https://red5pro.com).
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In an attempt to provide a more pleasing user experience and reduce data consumption on mobile devices, browsers are continuing to evolve their `autoplay` policies. While generally and attempt to keep websites (read: *ads*) from playing back unwanted and/or unsolicited video and audio, these policies also affect those sites in which the sole intent _is to_ playback video and/or audio - such as from a conference web application built utilizing [Red5 Pro](https://red5.net).
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Naturally, this can cause some confusion and frustration as `autoplay` may have worked as expected prior to latest browser updates. Thankfully, you do have options when using the *Red5 Pro WebRTC SDK* to provide a better user experience.
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package.json

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{
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"name": "red5pro-webrtc-sdk",
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"version": "13.0.0",
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"version": "14.0.0",
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"description": "Red5 Pro HTML SDK",
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"main": "red5pro-sdk.min.js",
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"scripts": {

red5pro-sdk.min.js

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