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.
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 TRTCis 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.
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).
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.
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. Not support pulling RTMP stream from TRTC; only support pushing stream to TRTC.
Primary domain is intl-rtmp.rtc.qq.com; backup domain is rtmp.rtc-web.com. If there are issues with the primary domain DNS resolution, you can use the backup domain.
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.
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 page and 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.
The screenshot below is an example of playback on the Web. You can also play the stream on other clients.
FAQ
Streaming failed
Common causes
No package purchased or expired.
Incorrect or expired signature.
Streaming with B-frames (appears as "stream ends after one second" on the dashboard); should set to baseline encoding.
Other causes
If the stream is pushed by an embedded hardware device, the URL may be truncated.
Streaming with H.265; should use H.264.
Setting chunk size too large on the client; recommend setting chunk size to 1360.
Lag and screen artifacts
Check the Tencent Cloud Real-Time Communication dashboard to monitor streaming frame rate stability. If stable, the issue is likely on the player's side - please investigate the player. If the frame rate is unstable, consider the following:
Check if the streaming client's local CPU and memory are under high load. If using OBS for streaming, observe the status bar at the bottom for information on dropped frames, network, CPU, and frame rate.
Check if the local network bandwidth is sufficient. Ping the streaming domain to observe RTT; use the network diagnostic tool to test the streaming domain and check bandwidth, ideally reaching 10M.
The streaming client may try reducing bitrate and frame rate to lessen client load, refer to the OBS settings in the main text, recommend setting bitrate to 1500 Kbps for 720p.
High Latency
If the client pulling stream uses TRTC anchor role, latency is usually lower than that of TRTC audience role. If not anchor role, change to that and compare to see if there is improvement.
Local encoding and network significantly affect the streaming end. Try different platforms for testing; if using OBS, consider streaming on a Windows system; ping the streaming domain to observe RTT.
Stream not visible on other ends
The streaming end used a string room number, but the pulling end used a numerical room number; modify the pulling end to use a string room number to enter the room.
Frequent disconnections and re-streaming
Username duplication, causing mutual kicking out; ensure that the userid is globally unique under a single sdkappid.
Streaming with B-frames; set to baseline encoding.
Server callback
An RTMP streaming user is also a user in the TRTC room, with no fundamental difference from other end users, see event callback.
Using your own domain
Configure your own domain CNAME to official domain - recommended for future use.
