CN1652560A - Comprehensive system of IP speech video frequency and PSTN telephone - Google Patents
Comprehensive system of IP speech video frequency and PSTN telephone Download PDFInfo
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- CN1652560A CN1652560A CNA2004100395125A CN200410039512A CN1652560A CN 1652560 A CN1652560 A CN 1652560A CN A2004100395125 A CNA2004100395125 A CN A2004100395125A CN 200410039512 A CN200410039512 A CN 200410039512A CN 1652560 A CN1652560 A CN 1652560A
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Abstract
With IP communication technique being adopted, the synthesized system makes a host running on IP protocol possible to connect to PSTN cordless telephone base station with 2.4G or 900 M through industrial standard USB interface or RS232 serial interface used in current. Moreover, data communication can be carried out from data in VOIP system to mobile station of cordless telephone through RF link by using the said interface. Multiple functional expansions can be carried out for the system, and pick-up head can be embedded inside mobile end. Mobile IP videophone terminal can be built through RF link. IP host may be specific set top box or PC possessing strong capability capable of down loading information such as address book etc. stored in IP host to mobile stations. Mobile stations can be multi-device system.
Description
Invention field
The present invention relates to the integrated telephony system of a kind of integrated PSTN and VOIP, belong to computer communication field.
Background of invention
In the world, adopt Circuit-switched PSTN, promptly public phone network still is the chief component of telephone network.VOIP utilizes packet technology to carry out the voice-and-data communication.Compare with PSTN, the integrated cost of VOIP is very low.Each voice channel speed of PSTN is 64Kbps, uses circuit to connect; VOIP can make actual voice flow minimum to from 2.4K to 16K owing to use voice compression technique.At present the most frequently used G.723.1 and G.729 two kinds of coded systems are respectively 5.3/6.3K and 8K (seeing the following form).
Standard code type bit rate MOS complexity time delay
G.711 PCM 64 4.3 1 0.125
G.726 ADPCM 32 4.0 10 0.125
G.728 LD_CELP 16 4.0 50 0.625
GSM RPE_LPT 13 3.7 5 20
G.729 CSA_CELP 8 4.0 30 15
G.723.1 ACELP 6.3 3.8 25 37.5
US Dod LPC-10 2.4 synthetic languages 10 22.5
The speech coding standard
Owing to use the mode of packet switch, can make the utilance of physical link be much higher than the circuit ways of connecting.Owing to these reasons, the integrated cost of VOIP is cheaper.In addition, the network of VOIP can extend further to the transmission network of video and text, and the PSTN network will realize that these functions can compare difficulty, perhaps can increase considerably cost.Owing to these reasons, the pstn telephone system upgrades at present, and in some developed countries, the VOIP network is progressively replacing PSTN.
In the pstn telephone system, service provider is available to be generally comprised basic voice and is engaged in, caller identification, and service such as fixed net short message.Because terminal node uses the analog line transferring voice, can only transmit very limited data message by DTMF/FSK, and can not guarantee very high accuracy rate.And VOIP is owing to be based on IP, digitized completely network, thus except voice, can provide abundant Additional Services, such as the Video service of high code stream, transfer of data such as literal picture, conference system or the like.And can allow the user select different service providers easily, and need not carry out any modification network.
In recent years, Internet development, instant messages system become a kind of brand-new communication modes.At present MSN, AOL are foremost instant messages system service providers; And the QQ of company of Chinese Tengxun is foremost domestic internet instant messages service provider.These instant messages systems have become people, especially youthful a kind of main contact and communication way.And these service providers also begin to provide voice service.The present invention also considers compatible this type systematic, and makes full use of the characteristic of this type systematic.
H.323 the main system of regulation VOIP call signaling comprises and SIP.
In order on the packet switching network of guaranteed qos not, to launch multimedia conferencing, by the 15th SG-15 of seminar of ITU in 1996 by the front page of suggestion H.323, and second edition was H.323 proposed in 1998.H.323 formulated the multimedia communications system standard on the packet network that no QoS (service quality) guarantees, these packet networks have been dominated current desktop network system, comprise Ethernet, Fast Ethernet, Token Ring, FDDI technology based on TCP/IP, IPX packet switching.Therefore, H.323 standard is used for LAN, WAN, Internet, multimedia communication on the internet technical foundation and guarantee is provided.
Overall, H.323 be a framework construction, it relates to the content of terminal equipment, video, audio frequency and transfer of data, Control on Communication, network interface aspect, has comprised that also multipoint control unit (MCU), multipoint controller (MC), multipoint processor (MP), gateway and the pass of forming multipoint conference equipment such as keep.
The audio coding algorithm that H323 adopts meets the ITU standard, and the encoder that is adopted generally has three kinds: ITU-TG.723 speech coder, ITU-T be speech coder, ITU-T speech coder G.729 G.728. and the major function of speech coder is exactly that PCM (pulse code modulation) sample value user speech is encoded into a spot of bit (frame).This method makes voice have robustness when link produces error code, network jitter and burst transfer.At receiving terminal, speech frame is decoded as PCM voice sample value earlier, and then converts speech waveform to.
Summary of the invention
The invention provides a kind of custom-designed integrated telephony system, can adapt to the applicable cases of the complexity in the communication network escalation process.The entire system structure chart is seen figure one.
The entire system structure
The present invention mainly comprises the system of the exchange of integrated treatment ip voice, IP video exchange, PSTN exchange of speech.System can adopt wireless communication technique, and voice are encoded, and sets up being connected of main frame and moving handle through the RF passage.From dividing in logic, system is divided into exchange control system and mobile system two parts.The main functional modules of exchange control system such as figure two.Mobile system functional module such as figure three and figure four are respectively the module of host side and cell phone end.
Exchange control system is handled call signaling, medium encoding and decoding, the switching of several paths etc.Call signaling comprises several standards of VOIP: H.323, and SIP etc.; Also comprise LOOPSTAR that PSTN comprises etc.What is more important because this system integrates VOIP and PSTN, must have complete call handover method.Following several situation is specifically arranged:
IP incoming call or IP breathed out when PSTN was not busy
PSTN incoming call or PSTN breathed out when IP was not busy
When PSTN hurries, the IP incoming call
When PSTN hurried, IP breathed out
When IP hurries, the PSTN incoming call
When IP hurried, IP breathed out
First two situation is simple relatively, as long as handle according to the handling process of standard; And, need to introduce some mechanism to other four kinds.Mainly be that conversation keeps and handover mechanism.Even function such as playing back music when also having conversation to keep.Several dispositions are described respectively below:
When PSTN hurries, the IP incoming call.At this moment the IP main frame is sent to the IP/PSTN switching system by USB with call request after treatment, and switching system mails to mobile system by the RF path with signaling.After mobile system is received, promptly produce specific incoming call prompting sound, this incoming call prompting sound and the PSTN sound mix of conversing.At this moment, mobile system provides several selections:
A. switch to IP, the PSTN maintenance of conversing
B. rejection IP incoming call continues the PSTN conversation
C. hang up PSTN, beginning IP conversation
When PSTN hurried, IP breathed out.In this case, following two kinds of possibilities are arranged:
A. hang up PSTN earlier, beginning IP breathes out
B. switch PSTN, beginning IP breathes out.This is that the PSTN conversation keeps
Two kinds of situations when IP is busy and above two kinds similar.
Audio coding and video coding relate to several standards.Generally speaking, the audio frequency of PSTN Network Transmission 64Kbps.VOIP be internet telephone system transmission be data through audio coding.Different coded formats has different baud rates; The IP main frame can be determined different coded formats according to different network bandwidth situations, and these will be realized in the capabilities exchange communication process that H.323 (or SIP) agreement is stipulated certainly.
For the PSTN system, can also realize fixed net short message and incoming call display function by FSK.Fixed net short message and caller identification are sent to moving handle by the RF path.
System can carry out the upgrading of VoIP software version by the IP main frame.After the IP main frame is connected on the Internet, whether software can have the version of renewal to issue by automatic server search to the backstage, if have, then carry out escalation process, be that the IP main frame sends the edition upgrading request to server, after request was admitted, the download path that the IP main frame provides according to server began to download new edition software.
The IP main machine structure
As figure two, the IP main frame mainly is made up of IP interface, voice-grade channel, video channel and USB (or RS232 interface).The IP main frame is connected to Internet by the IP interface.The IP interface can be the RJ45 of current popular, also can be to be connected to the Internet by isdn dialup.
The IP main frame explains and handles H.323 the agreement with SIP etc., and receives and send the coded data of Voice ﹠ Video respectively by the Voice ﹠ Video passage, and the IP main frame is also born the effect of audio/video encoding/decoding respectively.
Specifically, voice-grade channel receives the ip voice bag from the IP interface, according to different forms, selects different decoding adapters to decode.What obtain after the decoding is the PCM voice flow.Video channel receives the video data from the IP interface, the general video code model that uses H.263 and H.263+ define.What obtain after the decoding is RGB or IUV video dot array data.
The Media Stream that obtains behind the audio decoder sends by USB interface, and PSTN/IP exchange main frame can receive these data by USB interface.Can play by the video display system of IP main frame behind the video decode.
The IP main frame can be the PC with IP interface, also can be the special equipment with computing and network function.Internet streaming is exercised the MSN of usefulness at present, and ICQ or China domestic instant messages systems such as Tengxun all have the speech communication function, and PC, the PDA etc. that move these softwares can be as the part-structures of the IP main frame of mentioning among the present invention; Certainly except these softwares, also need to increase the drive software of USB, could be as complete IP main frame.Generally speaking, after the IP main frame is connected to the Internet, need Login Register to arrive special service system, just can become a node in the VOIP system, make a call and accept called.Each service system has nothing in common with each other to login process, but all should finish the function of authentication and terminal point information registration.
The another one function of IP main frame is the record address book.Because the IP main frame generally has abundant calculation function and memory space, so abundant address book function can be provided.Sophisticated functions in the various pstn telephones can very easily be set in addition, and, text-to-speech prefabricated such as the refusing to receive telephone number left a message or the like.Can also provide multiple pstn telephone not available function, show as the online user and can see which user can call out, as the communication of real-time voice and text can be provided simultaneously.
Exchange is generally USB being connected of main frame to the IP main frame with IP/PSTN, can certainly use RS232.If but system supports the video flowing exchange, RS232 can not provide enough bandwidth.
Exchange base structure
In figure two, PSTN exchange main frame and two modules of IP/PSTN switching system are integrated in the exchange base.That is to say the function of telephone exchange (TC) that the exchange base is integrated and IP/PSTN exchange.The TC module is very ripe, just repeats no more here.
The exchange base is connected PSTN exchange main frame and IP main frame by TC respectively with USB interface.Mention as previous section, in native system, total IP incoming call IP breathes out (PSTN spare time), PSTN incoming call PSTN breathes out (IP spare time), IP incoming call when PSTN is busy, when PSTN is busy IP breathe out, PSTN incoming call when IP is busy, several situations such as IP exhalation when IP is busy.
For Telephone set, during the IP incoming call, the IP main frame is by USB interface notice exchange base; And the PSTN incoming call can directly produce the system break of exchange base CPU.Which kind of mode no matter, Telephone set, is handled according to current state by predetermined logic.
Telephone set also includes the RF module, carries out communication with moving handle.The content of RF module communication comprises speech, video and literal.Telephone set sends speech that transmits from IP main frame or TC and the lteral data that transmits from the IP main frame by the RF link; Telephone set also receives the language and characters data that transmit from cell phone by RF, also has video data.
Mobile system
The RF module of Telephone set and mobile handset are the major parts that constitutes mobile system.In US and European country, 900MHz once was the main frequency range of the cordless telephone system in the mobile system, yet 2.4GHz is accepted as generally acknowledged free frequency range by countries in the world gradually in recent years.Basket tooth technology and 802.11 all is the wireless communication technique of 2.4GHz frequency range.Below explain the groundwork process of mobile system.
After cell phone (slave unit) enters in the radiation scope of base (main equipment), promptly receive query requests after, just begin to enter intermediateness.Through the communication process as figure two, master-slave equipment promptly enters connection status.After portable terminal entered in the query context of main frame, under connection status, if long-time not conversation, slave unit can saltus step be the low power consumpting state of dormancy, and this moment, equipment power dissipation was very low.When phone incoming call or exhalation, connection can be waken up and enter active state.Slave unit has two kinds of possible processes, and promptly system is in the wakeup process under called and two kinds of roles of caller.
When cell phone is breathed out or base receive that no matter ring up signal (base is called out handset after receiving call request) is from the PSTN passage or from the call request H.323 of IP tunnel incoming call, base and cell phone promptly begin to carry out wakeup process.Connect when being in active state, literal, voice and image can begin to transmit on radio channel.
Another function of mobile system is a base with current online IP contact person's download of information to cell phone, and like this, cell phone can be selected by simple, promptly begin to call out the online connection people.Concrete way is that cell phone is initiated request, notifies base need download the online connection people; After base is received, by USB interface notice IP main frame; Generally speaking, the timely bitcom of IP main frame operation all has the current online connection people's of record function, and the IP main frame is as long as issue base with these data by USB interface, and base is transferred to cell phone by wireless connections again.
Cell phone can insert camera module, and camera module generally has encoding function, and cell phone sends out through after the wireless chnnel coding these data by the RF link.This system can realize movably video conferencing system.
Description of drawings
Figure one is the entire system structured flowchart.As can be seen from the figure, system is made up of IP main frame, main station box travelling carriage.The IP main frame is connected on the Internet, and is connected on the Telephone set by USB interface.The IP main frame mainly is processing VOIP calling and functions such as voice, coding and decoding video, also carries out the exchange of voice and order by USB circuit and Telephone set.Telephone set is connected to the PSTN circuit by RJ11, and the pstn telephone base also comprises the RF module.Cell phone is connected with the base system by the RF link.The cell phone that different with cell phone in the common cordless telephone is among the present invention also comprises a camera module, can record a video and sends to Telephone set by the RF link, forms video-phone system.
Figure two is building-block of logic of the major part of this integrated system.As we can see from the figure, in fact the part of the Telephone set among the figure one has comprised PSTN exchange main frame and IP/PSTN switching system two parts.Wherein the IP/PSTN switching system has also comprised the RF module.The IP/PSTN switching system is responsible for handling various IP and the concurrent situation of PSTN.
Figure three is base system construction drawings of mobile system.The base system is mainly by CPU, multi-media signal processor, and baseband processor and RF module are formed.CPU is as the central processing system of base, and multi-media signal processor is handled from PSTN with from the voice signal of RF; Baseband processor, can send on the Radio Link by RF through after the baseband processor as through the processor after the source encoding.It is to be noted that usb interface module can provide the function that is connected with PC,, finish foregoing VOIP function by USB interface.Both can transmitting audio data by USB interface, also can the control channel signal.
Figure four is structure charts of the cell phone part in the mobile system.The cell phone system has camera, can support video to grasp.
What figure five showed is that mobile system wakes up and connection procedure.Not under the situation of within the main frame scope or not conversation, cell phone is in low power consumpting state at cell phone; Cell phone enters within the query context of main frame, promptly can be subjected to the request signal of main frame after, through the response of cell phone with call out and confirm etc. that cell phone is promptly set up with main frame and is connected, this connection status has different definition in different agreements.When the calling of PSTN or IP phone took place, mobile system was by similar process, and voice, video channel are opened respectively, enter into talking state from low power consumpting state, can begin conversation.
Claims (7)
- One kind be used for that integrated treatment IP internet call, PSTN are called out and handset between the integrated telephony system of calling out, it is characterized in that: comprehensive ip voice subsystem, IP subsystem of video, PSTN exchange of speech subsystem, with the wireless telecommunications subsystem, provide integrated communication system.
- 2. comprehensive ip voice subsystem according to claim 1, it is characterized in that: the ip voice call processing subsystem operates on the IP main frame, its major function is the audio call signaling (comprising H.323 and SIP) of processing VOIP, and carries out G.723.1, G.729 waiting the conversion of speech coding form.
- 3. ip voice call processing subsystem according to claim 2 is characterized in that: this subsystem receives the ip voice bag through overcompression, after the decompression, is sent to pstn telephone base end by USB interface again.This subsystem also receives the speech data of the uncompressed that transmits from PSTN base end by USB interface simultaneously, through behind the compressed encoding, sends on the Internet by the IP bag.
- 4. integrated telephony system according to claim 1 is characterized in that: the major function of IP subsystem of video is: handle IP video call signaling and coding and decoding video on the Internet; H.263+ and MPEG4 H.323 calling signaling comprise and SIP, and H.263 code/decode format comprise.Video code model can be upgraded as the plug-in unit of IP main frame.
- 5. integrated telephony according to claim 1 system is characterized in that: PSTN calls out subsystem and comprises that mainly this telephone circuit switching system of telephone circuit switching system of standard operates on the Telephone set.Comprise usb segment on the Telephone set, as the middle rotaring carrier of IP audio frequency and video.Telephone set also comprises the pstn telephone switching system simultaneously, and this base has IP and two passages of PSTN.Busy when a passage, when another passage had the incoming call request, base had conversation and keeps function, and provides prompting, and the user can be switched at IP and two passages of PSTN.
- 6. integrated telephony according to claim 1 system, it is characterized in that: the communication two party of wireless telecommunications subsystem comprises Telephone set and travelling carriage two parts.Telephone set is accepted the Internet Protocol telephone incoming call request from the IP main frame, also accepts the call request from the PSTN circuit, calls out travelling carriage by the RF path; Telephone set is also accepted the exhalation request of travelling carriage, and selects in USB and two passages of PSTN one according to the exhalation mode, calls out.
- 7. wireless telecommunication system according to claim 6 is characterized in that: the travelling carriage part can also and then with the IP main frame be carried out information exchange by RF passage and Telephone set.By these information exchanges, the address book information of IP main frame can be sent on the travelling carriage.
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| Application Number | Priority Date | Filing Date | Title |
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| CNA2004100395125A CN1652560A (en) | 2004-02-05 | 2004-02-05 | Comprehensive system of IP speech video frequency and PSTN telephone |
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| CNA2004100395125A CN1652560A (en) | 2004-02-05 | 2004-02-05 | Comprehensive system of IP speech video frequency and PSTN telephone |
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Cited By (8)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN101252614B (en) * | 2007-01-22 | 2011-09-28 | 索尼株式会社 | Communication apparatus, phone set, communication system, communication method |
| CN1968328B (en) * | 2005-11-14 | 2011-12-14 | 美国博通公司 | Voice communication method |
| US8274970B2 (en) | 2005-11-14 | 2012-09-25 | Broadcom Corporation | Voice communication device with PSTN and internet pathway analysis, selection and handoff |
| TWI406538B (en) * | 2005-11-14 | 2013-08-21 | Broadcom Corp | Pathways analysis and control in packet and circuit switched communication networks |
| CN102355516B (en) * | 2008-11-24 | 2014-04-02 | 中国电信股份有限公司 | Click-to-dial method and system based on fixed network, and client |
| CN105681167A (en) * | 2016-01-29 | 2016-06-15 | 深圳市泰比特科技有限公司 | Children smartwatch and wechat client talkback method |
| CN106992957A (en) * | 2016-01-21 | 2017-07-28 | 阿里巴巴集团控股有限公司 | Internet phone-calling is forwarded to the method and device of mobile communication terminal |
| CN107257422A (en) * | 2017-04-17 | 2017-10-17 | 太仓鸿策创达广告策划有限公司 | A kind of VOIP bimodulus gateway system |
-
2004
- 2004-02-05 CN CNA2004100395125A patent/CN1652560A/en active Pending
Cited By (10)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1968328B (en) * | 2005-11-14 | 2011-12-14 | 美国博通公司 | Voice communication method |
| US8274970B2 (en) | 2005-11-14 | 2012-09-25 | Broadcom Corporation | Voice communication device with PSTN and internet pathway analysis, selection and handoff |
| US8483100B2 (en) | 2005-11-14 | 2013-07-09 | Broadcom Corporation | Communication device supporting both internet and public switched telephone network telephony |
| TWI406538B (en) * | 2005-11-14 | 2013-08-21 | Broadcom Corp | Pathways analysis and control in packet and circuit switched communication networks |
| CN101252614B (en) * | 2007-01-22 | 2011-09-28 | 索尼株式会社 | Communication apparatus, phone set, communication system, communication method |
| CN102355516B (en) * | 2008-11-24 | 2014-04-02 | 中国电信股份有限公司 | Click-to-dial method and system based on fixed network, and client |
| CN106992957A (en) * | 2016-01-21 | 2017-07-28 | 阿里巴巴集团控股有限公司 | Internet phone-calling is forwarded to the method and device of mobile communication terminal |
| CN106992957B (en) * | 2016-01-21 | 2020-07-10 | 阿里巴巴集团控股有限公司 | Method and device for switching network call to mobile communication terminal |
| CN105681167A (en) * | 2016-01-29 | 2016-06-15 | 深圳市泰比特科技有限公司 | Children smartwatch and wechat client talkback method |
| CN107257422A (en) * | 2017-04-17 | 2017-10-17 | 太仓鸿策创达广告策划有限公司 | A kind of VOIP bimodulus gateway system |
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