A semantic integrity ensuring method under IP network environment

Abstract

This invention discloses a semantic integration insurance method in IP network, which comprises divide voice data send terminal by time length and seal voice data pack; it strictly buffer and play the voice data when the voice data pack amounts to certain number received; to substitute the loss or delay voice pack with noise data; to stop the voice play process of data receiving terminal when the number of the delay or loss packs exceeds certain threshold value; to await buffer data pack number amounts to certain delay number to start the voice play process.

Description

technical field [0001] The invention relates to a semantic integrity guarantee technology under the IP network environment, that is, a method for effectively ensuring the semantic integrity under the unreliable transmission protocol of the IP network. Background technique [0002] With the rapid development of computer network, the application of voice transmission system based on computer network technology is becoming more and more extensive. Due to the real-time nature of voice transmission, it has high requirements on the transmission rate and network conditions of the computer network. [0003] However, there are basically few computer networks that can meet such high requirements. On the other hand, the computer network communication channel is shared by multiple users. Channel sharing makes the computer network channel busy and idle, that is, the channel condition is good and bad, which further affects the performance and effect of the voice transmission system. [...

AI Technical Summary

Problems solved by technology

[0005] (1) The network time is extended, and the real-time performance of the system is poor

When the computer network is relatively busy, every node in the transmission process from the sending station to the receiving and sending voice data may wait for a long time because the node is relatively busy

Or because a certain section of the network is poor, you have to choose other routes with longer physical lines, which may cause a large delay in the arrival of data packets, which will cause the problem of poor real-time performance of the system. The voice of the sender may have been sent a long time ago

[0006] (2) Voice playback at the receiving end is not smooth

When the performance of the computer network is not stable, in a certain period of time, when the network condition is good, the time delay for the data packet to reach the receiving end may be very small; The delay may be relatively large. If it is not processed at the receiving end, intermittent voice will be generated during playback, resulting in unsmooth voice playback.

[0007] (3) When data is lost, the semantic integrity of the transmitted voice is poor

However, if there is a poor network situation, the voice played by the real-time playback system at the receiving end will be interrupted intermittently, which is contrary to the "data buffering method"

[0017] In addition, since the real-time playback system simply discards late data packets and replaces them with silence, when there is silence, the user is likely to mistake it for a pause in the speaker instead of data. Lost, may receive wrong voice

Obviously, semantic integrity cannot be guaranteed