From: http://www.edu.cn/20030616/3086964_2.shtml
Time delay and delay jitter
End-to-end delay includes codec delay, packaging and unpacking delay, and network transmission delay. Time delay changes, JITTER, is mainly caused by the network, and if the intermediate node (router, switch, etc.) passed in the end-to-end transmission path, the greater the time delay jitter.
Voice coding technology
Voice coding techniques can provide high quality speech while utilizing bandwidth. Different coding techniques will bring different speech quality, which lists several coding technology MOS (Mean Opinion Score) values (test results). Refer to the results listed in the table below, the G.729 and G.711 coding schemes can meet the quality requirements of the Education System IP Phone.
Coding Technology Bit Rate (K BPS) MOS Value Coding Delay (MS) G.711644.40.75G.723.1 (5.3K) 5.33.630g.723.1 (6.3K) 6.33.430G.72984.010MS GSM133.120
Coding technology and performance indicators
On the other hand, the bandwidth requirements of G.729 are much lower than G.711, in the same line quality and bandwidth, G.711 Datasheet is less than G.729, and this time is greater than speech compression Data reported time. Therefore, from the auditory angle analysis, the effects of the G, 729 and G.711 compression algorithms are basically the same.
Comprehensive consideration, G.729, which satisfies both voice effects, saving bandwidth, saving the investment cost of the line, is the preferred encoding compression algorithm.
Package loss rate
There are factors where IP packets are lost in IP networks are: packet loss in network transmission, network congestion, gateway devices actively packet. The voice quality will be severely affected when the package loss rate exceeds 10%.
echo
The echo is caused by the impedance mismatch between the two ends of the call. In the IP network in the end-to-end, the effects of echo interference are particularly obvious.
Voice level
Suitable speech transmission and reception levels are another important factor affecting the quality of the call, so the voice gateway must have the function of the voice level adjustment.
As can be seen from the above analysis, the voice quality to improve the IP phone should be mainly started from two aspects. First, use the appropriate speech coding compression technology to select a high quality speech gateway and appropriate gatekeeper control method; the second is to improve the quality of service (QoS) of the IP network, so that network delay, time delay jitter and package loss rate control That is within a certain limit.
The gateway is analyzed below, and the gatekeeper is guaranteed about the above factors.
Gateway guarantees measures for voice quality
Time delay and jitter are factors that affect speech quality vital, the gateway can be better controlled on both factors by the following technologies:
Silent suppression technology
Silent suppression, also known as the Voice Activity Detection, which is based on the voice and silence characteristics of people's day-to-day conversation, suppressed when the silence is detected, so that it does not occupy or minus channel bandwidth, detects When the activity speech is transmitted, it is compressed and transmitted. Studies have shown that the use of VAD technology can increase the effective utilization rate of channel bandwidth.
Jitter suppression buffer technology
The jitter suppression buffer is used to receive the end, and the purpose is to smoothly delay jitter, and take care of the decoding and compression operations.
Echo elimination
Measures to eliminate echo in IP telephony should be taken, and echo suppression is usually implemented in the gateway device.
