Colleagues who have really understood real-time audio and video development know that the technical reserves and skill requirements required for real-time audio and video development are relatively high. How far is it? In order to avoid the situation of insufficient testing and blind launch that leads to poor user experience, without more professional knowledge, tools, and equipment, how to send materials and use the simplest and most intuitive method to evaluate or What about evaluating a set of real-time audio and video solutions?
In real communication, we often encounter network freezes and unsmooth communication. The problems it reflects are likely to be problems such as high packet loss rate and insufficient bandwidth in real networks. So, without a professional network loss environment, how to quickly simulate and test the quality of audio and video calls under different packet loss rates and different bandwidth restrictions? Here we recommend that you use the iOS built-in network loss simulator to do simple packet loss and bandwidth limit tests. The specific steps are set in the following subsections.
Click "Settings" ---- After entering, slide to "Developer", click to enter
(Note: Regarding the developer options on iOS. To enable this feature, you need to connect your iPhone or iPad to a Mac computer, then open the Xcode development tool on the Mac, and then " Developer" option)
Click to enter "Status" (the default is off initially) ---- After entering, turn on "Enable" at the top of the menu (default is off)
Click "Add a profile..." to create a new test setting according to your own test requirements, and you can create a new name in "Name" for easy labeling
After the settings are complete, click "Save", and then you can perform packet loss or bandwidth limit tests as needed
As we all know, network bandwidth has always been very expensive, and the data transmission design of the current mainstream real-time audio and video solutions are all based on P2P. However, its ability to respond to the complex Internet environment is low, and the transmission quality is difficult to guarantee. When testing this kind of communication products, we often encounter such problems. When testing on the company’s intranet, it is smooth and clear, but once it reaches the user’s hands, the call becomes stuck and not smooth, or even difficult to make a call. Instant messaging chat software app development can add Wei Keyun's v: weikeyun24 consultation
In fact, the network conditions of different operators, regions, and mobile data types are very different, but in real applications, cross-operator, cross-region, and cross-communication network scenarios are very common. In the test before going online, this part of the risk is often easily overlooked.
In order to ensure the quality of the product, when testing an audio and video product, it is recommended to conduct a simulation test in a non-P2P network environment, for example, between different network operators (mobile, telecom, China Unicom) and different regions , Connect to different mobile data networks (2G, 3G, 4G) communication, etc. If overseas users and global applications are also considered, this part of the test needs to be paid more attention to.
In many cases, devices play a decisive role in the quality of audio and video, and this effect is easily overlooked by developers because it is not very intuitive. WebRTC's audio and video solution has some obvious problems on low-end android phones.
For example, if you use an external speaker to make calls on models such as Redmi 2A, Redmi Note 1 3G, and Huawei Honor 3C, there will be an inevitable echo. Another example is the inevitable current sound when the iphone6s is in the external state. This kind of problem is not easy to find, but it will have a great impact on the user experience after it appears.