Deploying a video chat system: testing network quality and deploying a video chat system

Deploying a video chat system: testing network quality and deploying a video chat system

For a real-time network voice and video system, the quality of the network plays a decisive role in the user experience of the system. Therefore, before the system is officially deployed, it is necessary to conduct comprehensive network testing and network optimization. This would be a complex system engineering, and it would be best if there is a professional team to do it. However, developers or implementers usually do these tasks. For example, the following analysis is required: Which cities are the target users in? Where or where the server is deployed (Distributed Solution) is most effective for the overall target users? How to deploy it? What is the bandwidth required? Do you need to support dual or multi-line connections (China Telecom, China Unicom, China Mobile, and China tietong )? And so on.

This article does not intend to systematically introduce these content, but only takes out the most important part. Small and medium enterprises without professional network optimization teams can follow the information given below, perform necessary tests and analysis. After completing these steps, I will be aware of the basic network conditions.

I. bandwidth usage

What are the network bandwidth requirements of voice, video, whiteboard, remote desktop, and other functions in a voice/video chat system or video conferencing system developed based on the OMCS voice/video framework?

Let's assume a common scenario: Assume that N online users perform one-to-one Multimedia Communication (I .e., divided into N/two groups) at the same time, without considering P2P channels, the following table lists the total bandwidth usage:

For video and remote desktop --

Frame Frequency: 8 ~ 10 fps.

Normal quality: the value of EncodeQuality is about 8.

High quality: the value of EncodeQuality is about 3.

Note:

1. Traffic Symmetry

For servers, the upstream and downstream traffic is symmetric; for clients, the inbound and outbound traffic is almost symmetric. The above table lists only one-way traffic.

2. Proportional Calculation

Take a video as an example. If the video size is not 320x240, the bandwidth usage can be calculated proportionally. If the video size is 640x480, the bandwidth usage will increase by 4 times (640x480)/(320x240 )).

3. Consider P2P

If the P2P channel is enabled, the server bandwidth usage decreases, but the client bandwidth usage remains unchanged. Assuming that the P2P success rate is 70%, the server's bandwidth usage will be reduced to 30% of the original.

4. Video Conferencing

The above data is based on one-to-one multimedia communication. If it is similar to a video conferencing scenario, the communication is many-to-many. At this time, the bandwidth usage will increase, the upstream and downstream traffic of the server is no longer symmetric.

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

 

II . Server shared bandwidth and exclusive bandwidth

Voice and video data are collected in real time and played in real time. In addition to the server bandwidth speed requirements, the stability of server bandwidth communication quality is also required, that is, low network latency and low network jitter. It is easy to understand that if the network jitter is large, the sound is intermittent (OMCS has a built-in jitter buffer JitterBuffer, but it can only alleviate this problem to a certain extent ).

Therefore, the server bandwidth must be dedicated bandwidth, and the shared bandwidth cannot meet the requirements of real-time voice and video. For real-time voice videos, m of shared bandwidth is not as dedicated as 5 m. This is why the IDC will send you m of shared bandwidth for free when you rent a server, while renting 5 m of dedicated bandwidth costs thousands of yuan a year.

Note:

(1) The bandwidth unit of the IDC server is bits/s, while the network speed unit is bytes/s. There is an eight-fold Relationship between them. For example, if the server bandwidth is 1 Mbps, the download speed can reach a maximum of kb/s.

(2) IDC server bandwidth refers to the sum of uplink and downlink bandwidth. For example, if the server bandwidth is 1 Mbps, the download speed and upload speed will not exceed kb/s at the same time.

 

3. Network quality testing and monitoring

1. Client Network Jitter

After the bandwidth quality of the server is guaranteed, if the clients involved in the voice and video sessions want a smooth experience, the following highlights are required:

(1) The ping latency from the client to the server is less than 100 ms.

(2) The maximum jitter range of ping cannot exceed 20 ms.

Network jitter has a greater impact on smoothness. During the test, we recommend that you open the ping to the server, so that you can observe the effect of ping on the smoothness of the voice video.

Note: The ping command can be continuously pinged by adding-t. For example, ping 192.168.0.123-t.

2. Observe network traffic

During the test, we recommend that you install NetLimiter network monitoring software on each client machine to view the upstream and downstream traffic between the client and the server in real time, as well as the network traffic on the P2P channel between the client and the client.

By combining network traffic monitoring and ping, you can easily test the real-time network status.

3. Test the network speed between the client and server

Using the copy file function of windows Remote Desktop and the NetLimiter monitoring above, we can easily test the network speed between the client computer and the server.

(1) On the client computer, connect to the target server by using the Remote Desktop function provided by windows (for example, Start Menu> All Programs> attachments> Remote Desktop Connection under windows 7.

(2) Uplink copy: copy a file larger than 50 MB from the current computer to the server.

(3) downstream copy: copy a file larger than 50 MB from the server to the current computer.

(4) Open the NetLimiter interface while the copy is in progress, and continuously observe the network speed transmitted between the client and the server.

(5) during the test, we recommend that you observe for more than 5 minutes. Note: (1) What are the uplink and downlink speeds? (2) is the speed stable?

(6) In a system similar to video conferencing, assume that four people are generally in the same meeting room, then four representativeness can be selected (the geographical region is representative) and then perform the test on the four computers at the same time, respectively recording the results of these four tests.

(7) During this test, we can observe the continuous ping value of the server at the same time.

Then, analyze each result to see if it can meet the bandwidth requirements of OMCS.

NetLimiter is as follows:

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