What factors affect the video and audio effects of the video conference system

In the application of the video conference system, the factors that affect the video and audio effects are mainly concentrated in three aspects:

1) Network service quality;

2) Performance of MCU and terminal;

3) Design of conference room.

1. Quality of service (QoS) of the network

At present, the network commonly used in the video conference system mainly includes E1 dedicated line and IP. E1 dedicated line is based on circuit switching and time division multiplexing technology, which can provide end-to-end exclusive bandwidth, so the network itself has a perfect transmission quality guarantee mechanism. In most cases, the main factor affecting the transmission effect of E1 dedicated lines is the quality of transmission equipment and transmission lines. For such factors, we can often improve by replacing the transmission equipment and reducing the line error rate.

The IP network is based on statistical multiplexing and packet switching technology. When it needs to simultaneously transmit multiple services such as voice, data, and video, its traditional "best effort" mechanism exposes many problems, the most important of which is that it cannot be This kind of service provides end-to-end bandwidth guarantee, which will cause large transmission delay and jitter. To this end, we must optimize the IP network through technical means to reduce the impact of the network itself on the effect of the video conference system. These technical methods have now developed into an important branch of the IP system, that is, quality of service (QoS).

The so-called QoS refers to the ability of a network to provide better service for a specific network traffic through multiple technologies. Its main purpose is to achieve priority control, including bandwidth, delay, jitter and packet loss. Almost all networks can take advantage of QoS to obtain the best efficiency.

QoS technology is divided into three categories, including best effort service, integrated service, and differential service, of which differential service is the most widely used. In differential services, the network classifies, queues, and manages data packets according to the QoS marking of each data packet. These tags can be IP addresses, TCP port numbers, or specific fields in IP packets.

In actual network planning, network devices (such as routers) are required to provide a QoS guarantee mechanism through a variety of technologies with the help of complex traffic management systems, and to classify different priority levels according to service types, such as optimal voice, video second, At the end of the data, network resources are then allocated according to these priority levels.

For video conferencing, in order to ensure the bandwidth of the video service, the router must be able to identify and classify the video service data packets in the passed IP data stream, and then provide bandwidth guarantee and priority delivery services through the congestion management mechanism. In this way, when the network is congested, the transmission effect of voice and video services can be guaranteed. At present, mainstream router manufacturers can provide QoS support based on classification, marking and congestion management.

2. Performance of MCU and terminal

In addition to the network should provide a good QoS guarantee mechanism, the video conference system equipment itself should also have good performance to truly ensure the effect of the conference. These performance factors include the video and audio codec technology used by the system, the design structure of the device, the ability of the device itself to adapt to harsh network environments, and other aspects.

1. Video and audio codec technology

Video and audio coding technology is a key technical indicator of the video conference system and an important factor that affects the conference effect. At present, the video coding technologies used in the video conference system mainly include H.261, H.263, H.264, MPEG-2, MPEG-4, etc., and the audio coding technologies mainly include G.711, G.722, G.728 , G.729, MP3, etc.

Among them, H.264 and MPEG-4 video encoding technologies can achieve high-definition dynamic image effects at low bandwidth, and the encoding delay is small. As a new generation of video encoding and decoding standards, its advantages are very obvious.

In terms of audio coding, MP3 is an efficient sound compression algorithm, its frequency response range is between 20Hz and 20KHz, the sampling frequency reaches 44.1KHz, and it supports two-channel coding, so it is getting more and more widely used.

2. The design structure of the equipment

In the early days, many MCUs and terminals in the video conference system used PC as the hardware structure, and the operating system was based on Windows. This type of equipment has great limitations in terms of codec performance, packet forwarding efficiency, stability, and security, resulting in low video and audio quality and large delays.

As a professional conference room application, most video conferencing systems now choose MCUs and terminal devices based on embedded design structures. This is mainly because the embedded system has streamlined instructions and high real-time performance, and combined with a dedicated codec DSP, it can realize high-quality, low-latency video and audio signal processing, and high stability and security.

3. The adaptability of the equipment to the harsh network environment

The QoS of the network can guarantee the transmission effect of the video conference to a certain extent, but its role is very limited, especially in some harsh network environments. The adaptability of the video conference system equipment itself to the harsh network environment will also have a greater impact on the conference effect. These adaptation capabilities include IP priority setting, IP packet sequencing, IP packet repetition control, IP packet jitter control, packet loss retransmission, and automatic rate adjustment.

1) IP Precedence

When planning the QoS technology of the differential service mode in the network, a variety of matching methods can be used to classify the service packets entering the data network, including IP addresses and IP precedence.

Among them, the use of IP priority in the IP packet can be used to prioritize audio, video and RTCP (Multicast) data streams. When the network uses IP Precedence for traffic matching, the video and audio packets with modified IP Precedence field information sent by the video device can be queued to ensure the priority transmission of the video conference stream.

2) IP packet sorting

Generally, the best-effort delivery mechanism of the network cannot guarantee the correct order of the data packets it forwards. For the H.323 video conference system, if the video equipment receives IP packets in sequence, it will cause a problem of out-of-sequence, and the loss or delay of the data packets will cause the freezing of the video image or the interruption or jitter of the sound.

This problem can be solved by the video device supporting the IP packet sequencing function. When the IP packet arrives, the video device will verify its order, and the out-of-order packets will be returned to maintain the continuity of the audio and video streams sent to the end user.

3) IP packet repeat control

An IP packet may generate multiple duplicate copies when passing through the bearer network, or in order to adapt to the harsh network environment, the system may also use the retransmission mechanism to generate multiple duplicate copies, which will cause freezing of the video image or sound interruption. Video equipment that supports repeated control of IP packets can use this function to correct the error to maintain the continuity of the audio and video streams sent to end users.

4) Jitter control

When the audio and video IP packets leave the sending end, they are evenly arranged at regular intervals. After passing through the network, this uniform interval is destroyed by different delay sizes, resulting in jitter. Jitter can cause discontinuities in the audio and video streams on the target terminal. Video equipment that supports jitter control can achieve jitter elimination through jitter buffering to maintain end user access