CN103428175A - Real-time voice intercom system based on heterogeneous mobile IP networks - Google Patents

The invention discloses a real-time voice intercom system based on complex heterogeneous mobile IP networks in the prior art. The system relates to connecting a set module, a P2P transmission module, an audio coding module, an audio decoding module and an audio playing and buffering module. Voice intercom software adopting the structural model can break limitations and achieve interconnection between heterogeneous mobile IP network types. Mainstream mobile IP networks at present includes: Wifi, EDGE, WCDMA, HSDPA, HSUPA and other 2G and 3G networks, point-to-point media stream transmission is achieved, dependency on server performance in a transmission client side/server (C/S) mode is reduced greatly, and instantaneity of voice intercom is improved. Advanced audio encoding and decoding algorithms are adopted in the audio coding/decoding module, compression on audio signals is carried out on the basis of guaranteeing audio quality, and therefore bandwidth is saved to the maximum. A self-adaptation voice buffering algorithm is adopted in the software, and therefore influences on tone quality by network delay variation in the system are eliminated.

Real-time voice intercom system based on the isomery mobile IP network

Technical field

The present invention relates to the mobile device data communication field, particularly Intelligent mobile equipment mobile IP network communication field.

Background technology

Traditional intercom, be all to use special-purpose hardware device, the coding/decoding capability of hardware is had relatively high expectations, and have that communication distance is short, intercom function is single, expensive, radiation has more greatly the shortcomings such as certain harm to health.Particularly higher to outside environmental requirement, other external signals are larger to its interference.

And there are the shortcomings such as power consumption is high, audio data coding efficiency is low, bandwidth occupancy is large, sound quality is poor in current existing Mobile phone talkback machine.Because being all adopts traditional client/server (C/S) pattern to carry out the transmission of audio data, thus large, poor to isomery mobile IP network compatibility to the server performance dependence, and can the real-time of voice data be exerted an influence.These all carry out intercommunication and have formed great restriction between different mobile IP networks to Intelligent mobile equipment.

Summary of the invention

The objective of the invention is to have overcome above-mentioned shortcoming of the prior art, designed and Implemented a real-time voice intercom system that can adapt to current isomery mobile IP network.

The technical solution used in the present invention is for this reason: the present invention includes to connect and set up module, P2P transport module, audio recording buffer module, audio coding module, audio decoder module and audio frequency play buffering module;

Wherein: connection is set up module and is obtained IP address separately for setting up associated two ends, then both is connected;

Described audio recording buffer module is converted into digital signal by the original sound typing and by the audio coding module by audio signal;

Described P2P transport module to the audio decoder module and be audio signal by it by digital signal decoding, then broadcasts by audio frequency play buffering module described digital data transmission at receiving terminal.

Described connection is set up module and is comprised address resolution module and interconnect module;

Wherein said address resolution module is obtained the IP address at associated two ends by public network server, the TCP/UDP that then utilizes described interconnect module to complete two ends connects.

Establish the transmission control protocol module in described P2P transport module.

Described audio frequency play buffering module is provided with jitter-buffer

The present invention is lower to hardware requirement, can be applicable to the equipment such as Android, IPhone, IPad of current main-stream, realizes the direct-connected communication system between distinct device.This system is not subject to communication distance restriction, it is efficient to have, without special hardware supports, data traffic is little, sound quality is high.

The accompanying drawing explanation

Fig. 1 is principle of the invention figure.

Fig. 2 is system block diagram of the present invention.

Embodiment

Real-time voice intercom system of the present invention has following formation:

The present invention includes to connect and set up module, P2P transport module, audio recording buffer module, audio coding module, audio decoder module and audio frequency play buffering module.Setting up module in connection is after two ends connect, the connection that the utilization of P2P transport module has been set up, and the audio data transmission that the audio coding module is produced is to opposite end.Opposite end is play through after audio decoder, again voice data being write to automatic adaptation cushion.

Connection of the present invention is set up module and is comprised address resolution unit and interconnection unit.At first address resolution unit obtains the public network IP address of oneself by public network server, then exchanges by public network server the public network IP address that obtains the other side, and after acquisition, the recycling interconnection unit completes the TCP/UDP establishment of connection.

Connection audio recording/playing module of the present invention is to be responsible for the module that original sound is operated in system.The audio recording module is responsible for the original sound typing and audio signal is converted into to digital signal, for transmission and the other side, plays.Playing module is responsible for receiving that decoded digital signal is converted into audio signal and is play.

Audio coding/decoding module of the present invention is to be responsible for voice data is carried out the module of coding and decoding.The audio coding module is encoded to voice data, and the data after coding are only 1/4 ~ 1/5 of former voice data.After the audio decoder module is received the data after coding, to its broadcasting of decoding.

P2P transport module application RTP of the present invention is transmitted data.Packets of audio data is carried out in order and transmitted, and the sequence number in packet allows the recombinate packet sequence of transmit leg of recipient, and sequence number also can be for determining suitable package location simultaneously.In this module, use RTCP Real-time Transport Control Protocol that the Data dissemination quality feedback information is provided, flow control and congestion control can be provided.

Audio frequency play buffering module of the present invention adopts jitter buffer to solve the network delay jitter problem.Realize the dynamic adjustment of jitter buffer, obtained voice call quality preferably.