CN101707593A - Conference system based on tree-shaped servers, PC client sides and telephone terminals - Google Patents
Conference system based on tree-shaped servers, PC client sides and telephone terminals Download PDFInfo
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- CN101707593A CN101707593A CN200910154228A CN200910154228A CN101707593A CN 101707593 A CN101707593 A CN 101707593A CN 200910154228 A CN200910154228 A CN 200910154228A CN 200910154228 A CN200910154228 A CN 200910154228A CN 101707593 A CN101707593 A CN 101707593A
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Abstract
The invention relates to a conference system based on tree-shaped servers, PC client sides and telephone terminals, which comprises a plurality of client sides based on PCs, and a plurality of client sides which are based on telephones and connected to a PSTN (public switched telephone network), wherein the client sides are connected to a wide area network (WAN) and connected with secondary servers by the WAN and routers; and the secondary servers are connected with root servers, and the PSTN is connected to a remote server by a gateway. In the invention, the root servers, the secondary servers and the remote server form a tree-shaped network, thus realizing terminal-to-terminal distribution of VoIP audio stream and PSTN audio stream. A PC client sends the audio stream by the secondary servers to the root servers and receives the audio stream selected by adjacent secondary servers, sound mixing is implemented at the client side of the PC, and a client side based on the telephone receives a sound mixing packet from the remote server which receives an audio data packet of an active speak client and combines with the sound mixing packet into one packet. The conference system reduces the delay from a terminal point to a terminal point, and ensures that a terminal user obtains high-quality audios; and the structure of the tree-shaped servers ensures that a conference has scalability.
Description
Technical field
The invention belongs to a kind of phone and VoIP mixing conference system, specifically, the present invention is for based on the user of PC with set up a kind of conference system of communication session based on the user of telephone terminal.
Background technology
Meeting is important component part of voice communication network.Audio conferencing is being different local people, and carrying out real-time communication each other provides a kind of facilitated method aspect to play important effect.Now, cheap PC, the network equipment, telecommunications, and the validity of correlation technique have greatly changed people's exchange way, thereby cause the human sharp increase that is connected with fhe global the Internet.This connection is to be finished by the Internet.The Internet is based on a common protocols external member, and this protocol suite helps computer to be connected to the Internet.Part common protocol external member is Internet protocol (IP), i.e. TCP/IP standard agreement.Its regulation IP packet is the information unit of passing through the internet, and the internet provides does not have the bag transmission service that connects and try one's best.IP also comprises the Internet control Message Protocol (ICMP) of processing controls and error message.
For many years, be widely studied by global public internet transferring voice.1996, H.323 International Telecommunications Union (International Telecommunication Association) passed through the Internet Protocol telephone standard and has come by IP network transferring voice (VoIP) as a kind of means.H.323 agreement has stipulated that minimum standard is logical IP transmission of voice, for example, and call setup and control.Meet H.323 that the equipment of standard can pass through the IP transferring voice, but this voice call quality can't be guaranteed.Therefore, people have developed the Conference server that is called multipoint control unit (MCU) and have sponsored meeting, and the participant uses based on the equipment of PC and Internet connection to central MCU, rather than the traditional telephone plant (PSTN) by the public service telephone network.
Recently, meeting begins agreement (SIP) and is developed.It is to be finished by the SIP working group that formulates the Internet engineering duty group (IETF).Session Initiation Protocol is a kind of signaling protocol, and it is widely used in setting up and closing the multimedia communication meeting, as voice and video by internet call.Session Initiation Protocol with better function than H.323, it provides calls out control and expanded function setting.It can be handled the basic setup function and upgrade service abilities, as calling transfer.
Along with telephone industry at agreement and equipment, comprise the legacy equipment of existence, as developing rapidly of phone and switching network, it is the user of PSTN that the telephone network of high infiltration will need the VoIP meeting that quality services can be provided.
So this requires Conference server can provide the support based on the user of packet and switching system.Therefore need propose to satisfy this demand based on the server architecture of PC and telephone terminal meeting.Chinese invention patent application CN100344140C (day for announcing is on October 17th, 2007) provides a kind of phone video conferencing system, it comprises main control unit, digital signal processing unit, internal bus and Ethernet interface and EI/ analog of telephone line access interface, Ethernet interface is in order to be connected to Ethernet, make the computer user can add the phone video conferencing system, EI/ analog of telephone line access interface is by the phone access module in the digital signal processing unit, landline telephone/mobile phone is linked into this phone video conferencing system, this invention uses telephone subscriber and computer user conveniently to participate in a conference, but there is following defective in it:
(1) at audio processing modules, it uses the sound mixing method of a simple addition and subtraction.This will cause overflowing of audio mixing value, thereby has influenced the quality of audio frequency, produces the audio frequency of the property understood difference.
(2) it has adopted the audio mixing algorithm of traditional decoding-audio mixing-coding, and this has greatly increased the load of the central processing unit (CPU) of audio processing modules, thereby has greatly reduced audio processing modules processing multi-user's ability.
(3) it has increased delay end to end.
Summary of the invention
Of poor quality in order to overcome the prior art sound intermediate frequency, cpu load weighs and postpones end to end the defective of length, the present invention proposes a kind of capable processing participant's distinct device such as PC, landline telephone and according to participant's distinct device, provides the conference system of corresponding voice data to every participant.
The technical solution adopted for the present invention to solve the technical problems is:
A kind of based on tree-like server, PC and telephone terminal conference system, this system comprises:
A plurality of clients based on PC, this client is received secondary servers or directly is connected root server by wide area network and router, and secondary servers is connected to root server;
A plurality of telephone plant terminals that are connected with PSTN by telephone wire, public switch telephone network is connected to remote server by gateway, root server, secondary servers has also formed the network that is tree-shaped shape;
Secondary servers is used to scan RTP RTP packet, obtains the energy of present frame from all audio streams of receiving, relatively obtains the client identifier of a corresponding N ceiling capacity through energy value, and N the audio stream of selecting sent to root server;
Root server, be used to scan RTP RTP packet, with audio stream of receiving and the energy that from the audio stream that secondary servers sends, obtains present frame, the process energy value relatively obtains the client identifier of a corresponding N ceiling capacity, and N selected audio stream sent to remote server;
Remote server, the energy that is used to receive all public switch telephone network audio streams and calculates present frame, the ENERGY E of the present frame of audio data stream can be calculated by following formula (1) and obtain:
In the formula, x[k] }
K=0 ..., L-1The input signal of expression audio data stream;
Obtain the calling customer terminal identifier of a corresponding N maximum energy value by the size that compares the E value, finally select in client and telephone plant client based on PC, remote server is encoded wherein, and the public switch telephone network audio stream is packaged into packet and sends root server with Real-time Transport Protocol, root server forms that a plurality of packets of audio data send to or sends to client based on PC by secondary servers, and remote server forms mixed audio stream and sends to the telephone plant client.
Described client based on PC also includes and is used to gather audio signal and carries out the audio mixing module that audio mixing is handled.
Described audio mixing module, be used for the voice data of choosing is sent into the jitter-buffer formation, send into the formation of speech frame buffering area through behind the decoder decode, regularly activate the audio mixing algorithm process, it is to extract the speech frame that arrives the earliest from every row speech frame buffering area that audio mixing is handled, do the voice signal audio mixing and handle, the audio mixing formula can be expressed as:
Wherein, input[j] [i] }
I=0 ..., N-1Represent the included speech frame of j row voice flow, the voice input signal that after decoding, obtains; Output result after audio mixing is handled is placed to mixing[i], M represents the sum of voice flow, and i represents the sample point index of audio data stream input signal, and N represents the size of speech frame.
Beneficial effect of the present invention mainly shows:
1, uses the audio stream selection algorithm, avoided the audio mixing algorithm of traditional decoding-audio mixing-coding, greatly reduced root server, the cpu load of secondary servers and remote server.
2, reduced the delay of end points to end points.
3, guarantee that the terminal use obtains high-quality audio frequency.
4. the framework of tree-like server guarantees that meeting has scalability.
Description of drawings
Fig. 1 is a conference system structure chart of the present invention.
S represents to select among Fig. 1; PS is expressed as part and selects; Mix is expressed as blender; Mux is expressed as multiplexing.
Fig. 2 is the detailed diagram of secondary servers.
Fig. 3 is the detailed diagram of root server.
Fig. 4 is the detailed diagram of remote server.
Embodiment
Come the present invention is further specified below in conjunction with specific embodiment, but do not limit the invention to these embodiments.One skilled in the art would recognize that the present invention contained in claims scope all alternatives, improvement project and the equivalents that may comprise.
Of the present invention based on tree-like server, the conference system of pc client and telephone terminal, this system comprises:
A plurality of clients based on PC, this client is received secondary servers or directly is connected root server by wide area network and router, and secondary servers is connected to root server; Described client based on PC is the computer of operation and internet, can carry out any communication session by Session Initiation Protocol.
A plurality of telephone plant clients that are connected with PSTN by telephone wire, public switch telephone network is connected to remote server by gateway, and root server, and secondary servers have formed the network that is tree-shaped shape;
Secondary servers is used to scan RTP RTP packet, and the audio stream of receiving is obtained the energy of present frame, relatively obtains the client identifier of a corresponding N ceiling capacity through energy value, and N the audio stream of selecting sent to root server;
Root server, be used to scan RTP RTP packet, with audio stream of receiving and the energy that from N the audio stream that secondary servers sends, obtains present frame, the process energy value relatively obtains the client identifier of a corresponding N ceiling capacity, and N selected audio stream sent to remote server;
Remote server is used for all the public switch telephone network audio streams that will receive and the energy that calculates present frame,
The ENERGY E of the present frame of audio data stream can be calculated by following formula (1) and obtain:
In the formula, x[k] }
K=0 ..., L-1The input signal of expression audio data stream;
Obtain the calling customer terminal identifier of a corresponding N maximum energy value by the size that compares the E value, finally select in client and telephone plant client based on PC, remote server is encoded wherein, and the public switch telephone network audio stream is packaged into packet and sends root server with Real-time Transport Protocol, root server forms that a plurality of packets of audio data send to or sends to client based on PC by secondary servers, and remote server forms mixed audio stream and sends to the telephone plant client.
Based on the client of PC, be the computer of operation and internet described in the present invention, can carry out any communication session by Session Initiation Protocol.
Described in the present invention based on the telephone plant terminal of PSTN, be meant legacy telephony equipment, can send a signal and start conversation to conference system of the present invention to PSTN, detect and demonstrate arrival by call arrival and data message, the instruction that receives conference system can start session, and transmission and reception are from the voice data of remote server.
Fig. 1 shows that the framework of conference system of the present invention and the connectedness of each part are the meeting of PC and telephone terminal.This conference system comprises that a gateway links PSTN, thereby is the traditional PSTN caller, has kept a traditional services cut-in method.
System architecture comprises numerous clients based on PC, and this client is connected to wide area network (WAN), receives secondary servers by wide area network and router.Secondary servers is connected to root server.In addition, system configuration also comprises numerous terminals based on phone, and these terminals are connected to the PSTN net.PSTN Netcom crosses gateway and is connected to remote server.In this framework, root server, secondary servers and remote server have formed the tree network that connects fully again, to realize VoIP audio stream and the distribution end to end of PSTN audio stream.
Client based on PC flows to root server and receives selected audio stream from contiguous secondary servers by the audio frequency that these secondary servers send them.Audio mixing is implemented in based on the PC customer.
Based on the terminal of phone, receive the audio mixing bag from remote server.
Described client based on PC also includes and is used to gather audio signal and carries out the audio mixing module that audio mixing is handled.
The audio mixing module, be used for the voice data of choosing is sent into the jitter-buffer formation, send into the formation of speech frame buffering area through behind the decoder decode, regularly activate the audio mixing algorithm processor, the audio mixing handling procedure extracts the speech frame that arrives the earliest from every row speech frame buffering area, do the voice signal audio mixing and handle, the audio mixing formula can be expressed as:
Wherein, input[j] [i] }
I=0 ..., N-1Represent the included speech frame of j row voice flow, the voice input signal that after decoding, obtains; I represents the sample point index of audio data stream input signal; N represents the size of speech frame; Output result after audio mixing is handled is placed to mixing[i], M represents the sum of voice flow; Here be set as 3.
The remote server audio mixing receives each client's who actively makes a speech packets of audio data and is combined into a bag.
In each secondary servers, we scan the Real-time Transport Protocol packet, obtain the energy of present frame from all audio streams of receiving.Relatively obtain three client identifiers of the energy of corresponding three maximums each other through energy value, Xuan Ding three audio streams send to root server like this.
At root server, we scan the Real-time Transport Protocol packet, from all audio streams of receiving with obtain the energy of present frame from the audio stream that secondary servers sends.Obtain three client identifiers of the energy of corresponding three maximums by the energy value that is compared to each other.Xuan Ding three audio streams send to remote server like this.
At remote server, we calculate the energy of present frame for all PSTN audio streams of receiving.The terminal identifier of three phones of the energy value of corresponding three maximums of acquisition.
Meanwhile, we carry out last selection based on the client of PC and three between based on the terminal of phone at three.Final selection result is listed in table 1.
Table 1: final selection situation
| ??VoIP | ??PSTN | |
| Case 1 | ??3 | ??0 |
| Case 2 | ??2 | ??1 |
| Case 3 | ??1 | ??2 |
| Case 4 | ??0 | ??3 |
Case 1: the notice root server is chosen three road VoIP stream.Root server forms a plurality of packets of audio data and sends to client based on PC.Remote server formation mixed audio stream sends to the terminal based on phone.
Case 2: the notice root server is chosen two-way VoIP stream.The encode audio stream of one road PSTN of remote server is packaged into it the Real-time Transport Protocol bag and sends to root server.Root server forms a plurality of packets of audio data and sends to client based on PC.Remote server formation mixed audio stream sends to the terminal based on phone.
Case 3: the notice root server is chosen one road VoIP stream.The audio stream of remote server coding two-way PSTN is packaged into them the Real-time Transport Protocol bag and they is sent to root server.Root server forms a plurality of packets of audio data and sends to client based on PC.Remote server formation mixed audio stream sends to the terminal based on phone
Case 4: the notice root server is not chosen VoIP stream.Remote server formation mixed audio stream sends to the terminal based on phone.Meanwhile, remote server uses pack this mixed audio stream and send to root server of Real-time Transport Protocol.Then, root server is passed to the client based on PC.
Claims (2)
1. conference system based on tree-like server, pc client and telephone terminal, this system comprises:
A plurality of clients based on PC, this client is received secondary servers or directly is connected root server by wide area network and router, and secondary servers is connected to root server; Described client based on PC is the computer of operation and internet, can carry out any communication session by Session Initiation Protocol.
A plurality of telephone plant clients that are connected with PSTN by telephone wire, public switch telephone network is connected to remote server by gateway, and root server, and secondary servers have formed the network that is tree-shaped shape;
Secondary servers is used to scan RTP RTP packet, and the audio stream of receiving is obtained the energy of present frame, relatively obtains the client identifier of a corresponding N ceiling capacity through energy value, and N the audio stream of selecting sent to root server;
Root server, be used to scan RTP RTP packet, with audio stream of receiving and the energy that from N the audio stream that secondary servers sends, obtains present frame, the process energy value relatively obtains the client identifier of a corresponding N ceiling capacity, and N selected audio stream sent to remote server;
Remote server is used for all the public switch telephone network audio streams that will receive and the energy that calculates present frame,
The ENERGY E of the present frame of audio data stream can be calculated by following formula (1) and obtain:
In the formula, x[k] }
K=0 ..., L-1The input signal of expression audio data stream;
Obtain the calling customer terminal identifier of a corresponding N maximum energy value by the size that compares the E value, finally select in client and telephone plant client based on PC, remote server is encoded wherein, and the public switch telephone network audio stream is packaged into packet and sends root server with Real-time Transport Protocol, root server forms that a plurality of packets of audio data send to or sends to client based on PC by secondary servers, and remote server forms mixed audio stream and sends to the telephone plant client.
Described client based on PC also includes and is used to gather audio signal and carries out the audio mixing module that audio mixing is handled.
2. according to claim 1 based on the conference system of tree-like server, pc client and telephone terminal, it is characterized in that described audio mixing module, be used for the voice data of choosing is sent into the jitter-buffer formation, send into the formation of speech frame buffering area through behind the decoder decode, regularly activate the audio mixing algorithm process, it is to extract the speech frame that arrives the earliest from every row speech frame buffering area that audio mixing is handled, and does the voice signal audio mixing and handles, and the audio mixing formula can be expressed as:
Wherein, input[j] [i] }
I=0 ..., N-1Represent the included speech frame of j row voice flow, the voice input signal that after decoding, obtains; Output result after audio mixing is handled is placed to mixing[i], M represents the sum of voice flow, and i represents the sample point index of audio data stream input signal, and N represents the size of speech frame.
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| CN103220258A (en) * | 2012-01-20 | 2013-07-24 | 华为技术有限公司 | Conference sound mixing method, terminal and media resource server (MRS) |
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Family Cites Families (5)
| Publication number | Priority date | Publication date | Assignee | Title |
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| US7218715B1 (en) * | 2000-08-28 | 2007-05-15 | Telefonaktiebolaget Lm Ericsson (Publ) | Method and system for setting up a telephone conference call in a switched telephone network through the internet |
| CN100442810C (en) * | 2002-12-23 | 2008-12-10 | 华为技术有限公司 | Mixed speech processing method |
| CN101252452B (en) * | 2007-03-31 | 2011-05-25 | 红杉树（杭州）信息技术有限公司 | Distributed type tone mixing system in multimedia conference |
| CN101465919B (en) * | 2007-12-19 | 2012-02-01 | 北京品视电子技术有限公司 | Method and system for implementing video conference |
| CN101227533B (en) * | 2008-01-31 | 2011-09-14 | 华为技术有限公司 | Apparatus and method for establishing audio conference connection |
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| CN103220258B (en) * | 2012-01-20 | 2016-07-27 | 华为技术有限公司 | Meeting sound mixing method, terminal and Media Resource Server |
| CN103037006A (en) * | 2012-12-24 | 2013-04-10 | 安徽华博胜讯信息科技有限公司 | Multilevel public electronic reading room traversing data pickup method |
| CN103037006B (en) * | 2012-12-24 | 2015-12-23 | 安徽华博胜讯信息科技有限公司 | Multilevel public electronic reading room traversing data pickup method |
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| CN112492255B (en) * | 2020-11-20 | 2022-07-05 | 杭州叙简科技股份有限公司 | Low-delay spanning tree audio and video conference method based on 5G |
| CN113271432A (en) * | 2021-06-30 | 2021-08-17 | 北京二六三企业通信有限公司 | Method and apparatus for transmitting and receiving speaker list |
| CN113271432B (en) * | 2021-06-30 | 2022-11-18 | 北京二六三企业通信有限公司 | Method and apparatus for transmitting and receiving speaker list |
| CN114500130A (en) * | 2021-12-30 | 2022-05-13 | 北京字节跳动网络技术有限公司 | Audio data pushing method, device and system, electronic equipment and storage medium |
| WO2023125350A1 (en) * | 2021-12-30 | 2023-07-06 | 北京字节跳动网络技术有限公司 | Audio data pushing method, apparatus and system, and electronic device and storage medium |
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