Network quality testing for the deployment of video chat systems

For a real-time network voice video system, the quality of the network plays a decisive role in the user experience of the system, so it is necessary to conduct more comprehensive network testing and network tuning before formally deploying the system. It will be a complex system engineering, if there is a professional team to do this thing is the best. However, a general company is a developer or an implementation person to do these things. For example, the following analysis is needed: Which cities are the target users mainly? Where or where (distributed scenarios) is the deployment server ideal for overall target users? How do I deploy it? How much bandwidth do you need? Is it necessary to support two or more lines (telecom, Unicom, Mobile, CTT, etc.)? Wait a minute.

This article does not intend to comprehensively systematically introduce these content, but only the most important part of it out, no professional network tuning team of small and medium-sized companies can follow the information given below to carry out some necessary testing and analysis. After doing this, the basic information about the network is roughly in the picture.

one . Bandwidth consumption size

In the Voice video chat system or video conferencing system, what are the requirements of voice, video, whiteboard, Remote Desktop and so on network bandwidth?

Let us first assume a common scenario: Assume that n online users at the same time 1 to 1 of multimedia communication (that is, divided into the N/2 group), regardless of the peer channel situation, the approximate bandwidth consumption is shown in the following table (with the Omcs voice video frame as an example, and QQ traffic requirements close):

For video and Remote Desktop--

Frame frequency: 8~10 fps.

Normal quality: The corresponding encodequality value is about 8.

High quality: The corresponding encodequality value is about 3.

Description

1. Flow symmetry

For the server, the upstream and downstream traffic is symmetrical, and for the client, the incoming and outbound traffic is almost symmetrical. Only one-way traffic is listed in the table above.

2. Proportional projections

Take video as an example, if the size of the video is not the same, then you can calculate the bandwidth consumption proportionally. Assuming that the video size is 640x480, the bandwidth will increase by 4 times times (640x480)/(+/-).

3. Consider the peer

If the peer-to channel is enabled, server-side bandwidth consumption is reduced, but client bandwidth consumption remains the same. Assuming that the success rate of peer-to 70%, the service side of the bandwidth consumption will be reduced to the original 30%.

4. Video Conferencing

The above data is based on 1 to 1 of multimedia communication, if it is similar to the video conferencing scene, then the communication is many-to-many, then, the bandwidth will increase the consumption of the server, the traffic is no longer symmetrical.

For example, there are M users chatting in a video room, each user's video will be broadcast to other (M-1) users, and each user to receive other (M-1) users of video data, so the bandwidth consumption will increase a lot.

two . Server shared bandwidth and exclusive bandwidth

Voice video data is real-time acquisition, real-time playback of data, in addition to the speed of server bandwidth requirements, but also requires the stability of the server bandwidth communication quality, that is, the network delay small, small network jitter. It is easy to understand that if the network jitter is large, the sound that is heard is intermittent (Omcs built-in jitter buffer jitterbuffer, but can only alleviate the problem to a certain extent).

Therefore, the bandwidth requirements of the server must be exclusive bandwidth, shared bandwidth can not meet the requirements of real-time voice video. for real-time voice video, 100M of shared bandwidth is not as exclusive as 5M. This is why when you usually rent a server, IDC will send you free 100M of shared bandwidth, and rent 5M of exclusive bandwidth, but spend thousands of dollars a year.

Also, be aware that:

(1)IDC Server Bandwidth Unit is BITS/S, and we usually say the speed of the unit is BYTES/S. It 's 8 times times the relationship between them. -for example, the server bandwidth is 1M, indicating that the download speed can reach about 120kb/s.

(2) IDC server bandwidth refers to the sum of upstream and downstream. For example, the bandwidth of the server is 1M, indicating that at the same time, the speed of download and upload speed together will not exceed 120kb/s.

three . Network quality testing and monitoring

1. Client Network jitter

After the bandwidth quality of the server is ensured, the clients participating in the voice video session, if they want to achieve a more fluent experience, need to achieve the following highlights:

(1) The client-to-server ping delay is less than 100ms.

(2) The maximum jitter range of the ping is not more than 20ms.

Among them, network jitter has a greater impact on fluency. When testing, it is recommended to open a ping to the server so you can observe the effect of ping on the fluency of the voice video.

Note: The ping command, plus-T, can ping continuously. such as Ping 192.168.0.123-t

2. Observing network traffic

When testing, it is recommended to install NetLimiter Network monitoring software on each client machine, which can view the traffic between the client and the server, and the network traffic on the peer channel between the client and client in real time.

By combining network traffic monitoring with Ping, it is easy to test the real-time status of your network.

3. Test the network speed between the client and the server

With the Remote Desktop Copy file feature from Windows, combined with the above netlimiter monitoring, we can easily test the network speed between the client computer and the server.

(1) On the client computer, connect to the destination server using the Remote Desktop feature that comes with windows, such as Remote Desktop Connection (Win7, Start menu, all programs, accessories, and so on).

(2) Upstream copy: Copy more than 50M files from the current computer to the server.

(3) Downstream copy: Copy more than 50M files from the server to the current computer.

(4) While the copy is in progress, open the NetLimiter interface and continuously observe the network speed that is passed between the client and the server.

(5) During the test, it is recommended to observe for more than 5 minutes, please pay special attention to the following: (1) What is the speed of the upper and lower lines? (2) is the speed stable?

(6) If it is similar to a video conferencing system, assuming that the demand is generally 4 people in the same conference room, then, you can choose 4 representative ( the geographical area of the representative) users, and then on these 4 people's computers to conduct this test simultaneously, recorded the 4 test results respectively.

(7) When you perform this test, you can observe the persistent ping value of the server at the same time.

Then analyze each result to see if it can meet the OMCS bandwidth requirement.

The NetLimiter is as follows:

Network quality testing for the deployment of video chat systems