AnyRTC update iteration in June, macOS added screen ID for screen sharing function, making sharing more efficient and simple; In addition, it solves the problem that the width and height of video is not 16:9, which leads to the loss of shared content. At the same time, it optimizes and improves a number of functions such as audio and video module and push stream component.

SDK

new

1. MacOS added screen ID for screen sharing

AnyRTC screen sharing is divided into two modes, screen sharing & area collection sharing and window sharing. The two modes of screen sharing can be applied to the sharing needs of all walks of life. Add screen ID for screen sharing, which can effectively solve the need of selecting screen sharing under multiple screens.

repair

1. Fixed video width and height not 16:9 causing up and down or left and right image reduction

When the self-collection function is used, the video source uses the screen to share the video content. Due to the inconsistency of the width and height ratio of the screen, the image will be clipped. This update fixes that any proportion of video stream data can be transmitted without clipping.

2. Fix data error when local push stream updates confluence information

Fixed the confluence error caused by data loss when the local push stream tool was doing confluence, and the layout could not be set according to the user’s setting parameters.

3. Fixed the flash problem of the local push stream component synthesis video

Fixed the problem that the video generated by the local push stream component flickered. When the anchor terminal called to update the layout, the video flickered intermittently, and the live broadcast could not be normal.

4, fix the Mac hardware coding rate is small problem

Fixed video blur when communicating with macOS using hardware encoding regardless of the bit rate set.

Upgrade the open source SIP gateway component

In order to make the landline and RTC interworking, anyRTC open source SIP and RTC interworking gateway, realize Web, Android, iOS, small program, SIP landline, PSTN telephone, mobile phone interworking. The gateway is used with two SDKS, RTC and RTM. RTM is responsible for signaling transmission and RTC is responsible for audio and video transmission. This upgrade supports multiple account configurations and custom messaging, such as the transfer of real phones or nicknames to the landing terminal.

Open source RTSP gateway upgrade

In order to monitor the cloud on the Intranet device with low latency requirement, anyRTC provides an open source RTSP to RTC gateway. The gateway consists of two modules: a pull component, which can pull the audio and video streams from THE RTSP, and a network transmission component, which transmits the audio and video streams obtained from the RTSP. The monitoring end can integrate anyRTC SDK to monitor the Intranet in real time.

The main features of the update include a fix for video hacking during disconnection and support for multitasking. Updated version before only shows the configuration switch a RTSP RTC function all the way, if you want to configure multiple flow, need developers themselves integration, development threshold is relatively high, using complex, updated developers only need in the configuration file for multitasking configuration, a key to start the script, greatly reducing the threshold of the developers and the use of the difficulty.

Open source interactive video demo

In order to promote the fast landing of video interactive lian-mic scene, anyRTC open source sample demo, sample demo shows three interactive lian-mic live form, developers can choose the mode according to their own scene.

  • Real-time live broadcast mode: the anchor, the audience and the audience all use real-time audio and video, and the delay can be controlled around 200ms.
  • Server side bypass streaming mode: The interaction delay between the anchor side and the audience is 200ms. The anchor side calls the server side bypass streaming mode, the server side broadcasts the interactive content in the form of CDN, and the audience side pulls the audio and video streams of HLS/HTTP-FLV/RTMP.
  • Client-side bypass streaming mode: The interaction delay between the anchor end and the continuous mic audience is 200ms. The anchor end deploys the local streaming component to push the interactive audio and video streams to CDN, and the audience end pulls the audio and video streams of HLS/HTTP-FLV/RTMP.

Github open source address :VideoLive