AnyRTC SDK is new in May, adding a variety of encryption types, so that real-time audio and video communication is more secure; Added 1080P resolution support for mobile push stream; In addition, several functions such as event reporting, log details, data statistics and network transmission have been optimized and improved.

The following is a summary of the updated content:

  • Mobile streaming supports 1080P

  • Reconstruct audio and video encryption and decryption to make communication more secure

  • Optimize log content and event reporting, facilitating fault locating

  • Optimize data statistics to make call details more intuitive and reliable

  • Optimize network transmission and dynamically adjust redundant packet sending

  • Optimized device management, supporting hot swap and rights management

  • Optimized Mac RTC library adaptation

  • Web RTC SDK browser adaptation optimized

  • Fix self-subscribe or muteRemoteStream without video issues

  • Fixed not being able to set roles in non-live mode

  • Fixed network quality judgment condition, network quality more accurate

  • Fixed error reporting of events

Here are the details of this month’s update:

new

1, mobile terminal push stream support 1080P

The new VERSION of the SDK API adds private methods to increase the mobile push stream from the original maximum of 720P to 1080P. The specific usage is as follows:

// Private method content
NSDictionary *parameter= [[NSDictionary alloc] initWithObjectsAndKeys:@"SetVideoMobile1080P"The @"Cmd",[NSNumber numberWithBool:YES], @"Enable",nil];
// Private method Settings
[_rtcKit setParameters:[ARCommon returnJSONStringWithDictionary:parameter]];
// Other audio and video configuration items.// Join the channel[_rtcKit joinRTC...] ;Copy the code

The preceding example uses iOS as an example

Private method calls must be set up before joining the channel

2. Reconstruct audio and video encryption and decryption to make communication more secure

The new version reconstructs audio and video encryption and decryption to significantly improve performance. The SDK uses AES128XTS encryption mode by default, and supports AES128ECB, AES256XTS, and SM4128ECB customized encryption modes. Students who need security reinforcement can continue to pay attention to anyRTC official website.

To optimize the

1. Optimize log content and event reporting to locate problems more easily

The new version makes log output positioning for each method and callback method. Through the log, you can know the order of SDK API call and the result of API call in detail. At the same time, critical events are reported. Developers can visually monitor the USE of SDK through the monitoring hall, and quickly locate problems in communication.

2. Optimize data statistics to make call details more intuitive and reliable

The new version of the data statistics has been deeply optimized to solve the problem of inaccurate statistics and incorrect reported data. Developers can intuitively give the client friendly tips through data statistics.

3. Optimize network transmission and dynamically adjust redundant packet sending

Audio and video transmission optimization dynamically adjusts the sending of redundant packets, reduces the consumption of network bandwidth by redundant packets, and reduces the performance overhead, so as to improve network utilization and transmission efficiency.

4. Optimized device management, supporting hot swap and rights management

When Windows gets the device list, the default device is placed first in the list.

Web SDK Obtaining device list Add parameters for applying for media device permission of the coarse browser. Developers can obtain parameters based on their own scenarios. Audio and video communication can be performed on devices without microphones or video collection devices.

5. Mac RTC library adaptation optimization

Mac platform RTC SDK library adaptation, to meet the developers in Mac native development requirements.

6. Web RTC SDK browser adaptation optimization

  • Safari on PC cannot communicate with audio
  • The Safari video window on the PC cannot play automatically
  • H5 video cannot be played automatically

Deep adaptation for mobile browsers, please check which browsers anyRTC Web SDK supports for details.

repair

  • Fix self-subscribe or muteRemoteStream without video issues

  • Fixed not being able to set roles in non-live mode

  • Fixed network quality judgment condition, network quality more accurate

  • Fixed error reporting of events

The above is the main content of SDK iteration this month, more detailed documents and anyRTC SDK historical updates, you can enter anyRTC official website – Developer Center “to view.