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Push Media Stream into TRTC

Last updated: 2024-08-12 17:57:06

    Overview

    Watch together, listen together, play together, learn together… Various experiences that once required face-to-face interaction are now moving online. Even if separated by thousands of miles, friends can still watch movies, listen to music and chat together. This amazing real-time interactive experience is becoming popular among young people and is now a major feature and mainstream direction of audio and video products.
    TRTC offers two streaming solutions: push online media stream and RTMP streaming with TRTC, each with their own application scenario, as detailed below:
    Push online media stream is used to pull cloud-based online media streams (online streaming or cloud-based on-demand files) and push them into a TRTC room.
    RTMP streaming with TRTC is used to stream local media files and audio and video from capture devices into a TRTC room via the RTMP standard protocol.
    Note:
    Relevant fees are as follows:
    Feature unlocking: Push online media stream and RTMP streaming with TRTC features need to be unlocked by a subscription to the basic or professional version of RTC-Engine monthly package.
    Usage fees:
    Using the streaming feature involves transcoding operations, incurring transcoding costs. For more details, see the Description of Billing of MixTranscoding and Bypass Relay.
    The costs of audio duration incurred by the streaming robot in the room are charged (Note: The costs incurred by the robot in the room for push online media stream features will not be charged by August 15, 2024, and they will begin to be charged on August 16, 2024).
    The audience in the room subscribing to audio and video streams will incur audio and video call costs. For details, see the Description of Billing of Audio and Video Duration.

    Push Online Media Stream

    Application Scenario

    Scenario type
    Description
    AI interactive classroom
    Relying on TRTC's ability to push online media stream, the platform enables online live interactive teaching by combining recorded real-life teaching videos with AI technology. This significantly reduces operational costs while ensuring teaching effectiveness. Before class, the platform records video segments for explanation of knowledge points, interactive questioning, feedback on questions, and answers based on the teacher's course setup, and uploads them to the video library. During the class, these videos are streamed to the TRTC room for live broadcasting through the TRTC push online media capability. Students can engage in interactive learning through voice and touchscreen. The server uses AI technology to evaluate real-time voice and responses of students, seamlessly switches teaching segments, and provides different real-time feedback, thus offering a personalized teaching experience.
    "Watch together" room service
    Live streaming contents, such as game live streaming, fashion shows, and sports events, can be pushed to a TRTC room through the TRTC's push online media stream capability for ultra-low latency synchronized viewing within the room. With TRTC's real-time interaction capability, the audience can communicate in real-time, cheer together, and enjoy an immersive viewing experience. On-demand programs like movies and music can also be inputted into the TRTC room through the capability, allowing users to share in real-time and chat with friends while watching.

    Feature Architecture

    1. Users create push online media stream tasks using the REST API. These tasks are executed by the relay server.
    2. The relay server pulls online streams or on-demand files.
    3. The relay server pushes the fetched audio and video to the TRTC room and automatically generates a virtual anchor user. The username and room number of this user are specified when the task is created.
    4. Other TRTC clients can watch these streams and utilize TRTC capabilities such as recording and relay.
    

    Feature Description

    The feature description of push online media stream is as follows:
    Type
    Description
    Task initiation method
    Users can initiate push online media stream tasks via the REST API. The audience can watch these streams, and features such as recording and relay are supported.
    Multiple source protocols and formats
    Protocols: HTTP, HTTPS, RTMP, HLS
    Formats: FLV, MP3, MP4, MPEG-TS, MOV, MKV, M4A
    Video encoding: H.264, VP8
    Audio encoding: AAC, OPUS
    Server-side callback
    When a push online media stream task is created and completed, it can be called backed to the server on the service side for service logic purposes. For detailed push online media stream events, go to view.

    Related Rest API

    Start push online media stream: StartStreamIngest
    Stop push online media stream: StopStreamIngest
    Query push online media stream: DescribeStreamIngest

    RTMP Streaming With TRTC

    TRTC supports the streaming of local media files and audio and video from capture devices into a TRTC room via the RTMP standard protocol. To facilitate your integration of TRTC, you can install OBS, FFmpeg, or other RTMP libraries to realize streaming with TRTC. OBS is a third-party open-source tool for live streaming. It is easy to use and free of charge, and it supports OS X, Windows, and Linux. OBS can be used in a wide range of scenarios and is capable of meeting most live streaming needs without requiring additional plugins. You can download its latest version from the OBS website.

    Use Cases

    Scenario
    Description
    Online education
    Use the desktop edition of OBS or FFmpeg to publish learning materials (most media formats are supported) over RTMP to a TRTC room. Students in the room can play the stream via the TRTC SDK and see the same learning materials as the teacher controls the playback progress/speed or switches between chapters. Excellent synchronization across multiple devices ensures better teaching quality.
    Sports watching
    Sports event organizers provide content in the form of RTMP streams. You can publish the streams to TRTC rooms so that users in the rooms can watch the event with ultra-low latency. With TRTC’s interaction capability, users can also audio/video chat with each other throughout an event.
    Others
    You can also use the RTMP publishing feature to implement other real-time interactive applications based on streaming.

    Architecture

    An RTMP client is a module of TRTC and can communicate with other TRTC clients. The interconnection delay is less than 600ms under normal circumstances. It can also use TRTC capabilities such as recording and relaying. The network architecture is shown in the figure below.
    

    Publishing and Playback URLs

    Publishing URLs

    rtmp://intl-rtmp.rtc.qq.com/push/Room ID?sdkappid=Application ID&userid=User ID&usersig=Signature
    For RTMP publishing, appName is push.
    Replace "Room ID", "Application ID", "User ID", and "Signature" with their actual values.
    For the sake of simplicity, we support only string-type room IDs. A room ID can contain numbers, letters, and underscores and cannot exceed 64 characters.
    Warning:
    To play an RTMP stream on other TRTC clients, when entering the room, make sure you use a string-type room ID.
    For how to generate UserSig, see UserSig. Make sure your signature is within the validity period.
    Example:
    rtmp://intl-rtmp.rtc.qq.com/push/hello-string-room?sdkappid=140*****66&userid=******rtmp2&usersig=eJw1jdE********RBZ8qKGRj8Yp-wVbv*mGMVZqS7w-mMDQL

    Usage Example

    You can use software or a programming library that supports RTMP to publish RTMP streams. The section below shows you how to do this.

    Using OBS to publish streams

    Prerequisites
    You have installed OBS.
    Step 1. Select a source
    In the Sources panel at the bottom, click +, and select a source based on your needs. Common sources include the following:
    Source
    Note
    Image
    Publishes a single image
    Image Slide Show
    Publishes multiple images (you can determine the order of playback and whether to loop the playback)
    Scene
    Inserts an entire scene to enable various streaming effects
    Media Source
    Uploads a local file and publishes it as a live stream
    Text
    Adds real-time text to your stream
    Window Capture
    Captures and publishes the window you select in real time
    Video Capture Device
    Captures and publishes the images captured by a camera in real time
    Audio Input Capture
    Audio live streaming (audio input device)
    Audio Output Capture
    Audio live streaming (audio output device)
    
    Step 2. Set publishing parameters
    1. In the Controls panel at the bottom, click Settings.
    img
    
    2. Click Stream and select Custom for Service.
    3. Enter rtmp://intl-rtmp.rtc.qq.com/push/ for Server.
    4. Enter a stream key in the following format:
    Room ID?sdkappid=Application&userid=User ID&usersig=Signature
    Replace "Room ID", "Application ID", "User ID", and "Signature" with the actual values, for example:
    hello-string-room?sdkappid=140*****66&userid=******rtmp2&usersig=eJw1jdE***************ZLgi5UAgOzoMhrayt*cjbmiCJ699T09juc833IMT94Ld7I0iHZqVDzvVAqkZsG-IKlzLiXOnEhswHu1iUyTc9pv*****D8MQwoA496Ke6U1ip4EAH4UMc5H9pSmv6MeTBWLamhwFnWRBZ8qKGRj8Yp-wVbv*mGMVZqS7w-mMDQL
    
    Step 3. Configure the output
    Because RTMP does not support B-frames, set the video encoding parameters as follows to remove B-frames.
    1. Go to Controls > Settings > Output.
    2. Select Advanced for Output Mode. The recommended Keyframe Interval is 1 or 2. For CPU Usage Preset, select ultrafast. For Profile, select baseline. For Tune, select zerolatency. And for x264 Options, enter threads=1, and then click OK.
    Warning:
    You need to remove the B-frames in RTMP streams, otherwise the connection will be disconnected after pushing. To achieve this, select baseline for Profile.
    
    Step 4. Set video parameters
    You can set video resolution and frame rate under the Video section of Settings. Resolution determines the clarity of video shown to audience members. The higher the resolution, the clearer the video. Frame rate (frames per second) determines playback smoothness. Typical frame rate falls in the range of 24 fps to 30 fps. Playback may stutter if frame rate is lower than 16 fps. Video games require higher frame rate and tend to stutter at a frame rate lower than 30 fps.
    
    Step 5. Configure advanced settings
    To reduce end-to-end delay, we recommend you do not enable Stream Delay.
    Keep Automatically Reconnect enabled and make Retry Delay as short as possible so that the publisher can be reconnected quickly after a disconnection occurs due to network jitter.
    
    Step 6. Publish the stream
    1. In the Controls panel at the bottom, click Start Streaming.
    
    2. If streaming is successful, the bottom bar will show streaming statistics, and the entry of a user will be recorded by the TRTC monitoring dashboard.
    
    Step 7. Play the stream on other clients
    As mentioned above in Set publishing parameters, to play the RTMP stream on other TRTC clients, you need to use a string-type room ID when entering the room. The screenshot below is an example of playing the RTMP stream on Web. (ps : You can go to the Demo pageand enter the room on any client-side to view the stream .)
    

    Using FFmpeg to publish streams

    To publish RMTP streams using FFmpeg commands or other RTMP libraries, you need to use the full URL, the H.264 video codec, and the AAC audio codec. For the container format, FLV is recommended. For GOP, one or two seconds is recommended. The configuration of FFmpeg parameters varies with different scenarios, so you need to have some knowledge of FFmpeg in order to use it to publish streams. The table below lists some common FFmpeg commands. For more options, see FFmpeg documentation.
    FFmpeg commands
    ffmpeg [global_options] {[input_file_options] -i input_url} ... {[output_file_options] output_url}
    Common FFmpeg options
    Option
    Note
    -re
    Reads input at the native frame rate. This is usually used to read local files.
    Options for output_file_options include:
    Option
    Note
    -c:v
    The video encoding library. libx264 is recommended.
    -b:v
    The video bitrate. For example, 1500k means 1,500 Kbps.
    -r
    The video frame rate.
    -profile:v
    The video profile. If you set it to baseline, B-frames will not be encoded. The TRTC backend does not support B-frames.
    -g
    The GOP (keyframe interval).
    -c:a
    The audio encoding library. libfdk_aac is recommended.
    -ac
    The number of sound channels. Valid values: 1, 2.
    -b:a
    The audio bitrate.
    -f
    The container format. Set it to flv. The FLV container format is required to publish to TRTC.
    Below is an example of reading a local file and publishing it to TRTC (note that quotation marks are required for the URL):
    ffmpeg -loglevel debug -re -i sample.flv -c:v libx264 -preset ultrafast -profile:v baseline -g 30 -sc_threshold 0 -b:v 1500k -c:a libfdk_aac -ac 2 -b:a 128k -f flv 'rtmp://intl-rtmp.rtc.qq.com/push/hello-string-room?userid=rtmpForFfmpeg&sdkappid=140xxxxxx&usersig=xxxxxxxxxx'
    Playback on other clients
    The screenshot below is an example of playback on the Web. You can also play the stream on other clients.
    
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