CN204350045U - A kind of signal transmission apparatus of IP phone - Google Patents
A kind of signal transmission apparatus of IP phone Download PDFInfo
- Publication number
- CN204350045U CN204350045U CN201420868288.XU CN201420868288U CN204350045U CN 204350045 U CN204350045 U CN 204350045U CN 201420868288 U CN201420868288 U CN 201420868288U CN 204350045 U CN204350045 U CN 204350045U
- Authority
- CN
- China
- Prior art keywords
- chip
- audio
- module
- phone
- signal transmission
- Prior art date
- Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
- Expired - Fee Related
Links
Abstract
The utility model relates to a kind of IP phone signal transmission apparatus, needs the CPU of very high primary frequency to cause waste resource to complete voice call to solve mobile phone A PP, and function is difficult to expansion, the problem that voice are not good enough.The signal transmission apparatus of IP phone, comprises Ethernet microcontroller, noise reduction chip, speech chip, master chip, LCDs, Switching Power Supply, step-down chip etc. containing audio coding module and audio decoder module; And, audio collection is connected by data wire with the audio frequency coding and decoding module of output module with master chip, master chip is connected with ethernet controller by Peripheral Interface, is connected with speech chip by Audio Data Line, is connected with LCDs by general-purpose interface.
Description
Technical field
The utility model relates to a kind of IP phone method for transmitting signals and the equipment for implementation method.
Background technology
Along with the demand of people's speech communication increases considerably, the transmission of voice by telephone wire, develop into Internet Transmission and become a kind of trend.
Although market has occurred mobile phone A PP can realize the Signal transmissions of IP phone, transmission, all based on operating system, has namely required higher to CPU frequency.For the Signal transmissions of the only simple call function of demand, the CPU very high by one piece of dominant frequency wastes resource, and each auxiliary products are difficult to interconnect, and function is difficult to expansion, and voice quality is not good enough.
Summary of the invention
The technical problem that the utility model solves is: provide a kind of not based on the method for transmitting signals of the IP phone of operating system and need the CPU of very high primary frequency to complete voice call to cause waste resource to solve mobile phone A PP for the custom-designed equipment of implementation method, each auxiliary products difficulty interconnects, function is difficult to expansion, the problem that voice quality is not good enough.
The technical scheme that the utility model adopts is as follows:
A kind of IP phone method for transmitting signals, comprise step: the registrar of transmission both sides invites checking, call response, the collection of voice signal, the coding of audio signal, RTP voice data package, UDP exhalation datagram transmission, the response of protocol stack module and UDP transport layer, protocol stack module is to the control of audio collection, protocol stack module controls the selection of coder module, protocol stack module controls the selection of decoder module, the control that protocol stack module exports audio frequency, the response of protocol stack module and UDP transport layer, UDP incoming call datagram transmission, RTP voice data unpacks, the decoding of audio signal, the output of voice signal.
Wherein, described audio signal sample, output adopt FM1188 noise reduction chip 101,201,301 and CS42L52 speech chip 102,202,302 to carry out.
Further, described audio frequency coding and decoding adopts the audio coding module 103 of G.711 algorithm, audio decoder module 104 carries out.
Further, the transmission of described udp protocol realizes being completed by W5200 ethernet controller 213,313.
Further, what described protocol stack module adopted is SIP/SDP agreement.
A signal transmission apparatus for IP phone, comprises Ethernet microcontroller, noise reduction chip, speech chip, master chip, LCDs, Switching Power Supply, step-down chip etc. containing audio coding module and audio decoder module; And, audio collection is connected by data wire with the audio frequency coding and decoding module of output module with master chip, master chip is connected with ethernet controller by Peripheral Interface, is connected with speech chip by Audio Data Line, is connected with LCDs by general-purpose interface.
Wherein, noise reduction chip is made up of FM1188 noise reduction chip 101,201,301, and speech chip is made up of CS42L52 speech chip 102,202,302.
Further, master chip is made up of the STM32F401 210,310 of built-in audio coding module 103, audio decoder module 104.
Further, ethernet controller is made up of W5200 ethernet controller 213,313.
The beneficial effects of the utility model are: the transmission equipment forming IP phone on one piece of bare board, use the CPU of lower dominant frequency to realize IP phone function, maximum saving resource; The communication of this IP phone method for transmitting signals adopts SIP/SDP agreement, and various corresponding product is easy to interconnect, and function is also easily expanded; Adopt G.711 agreement, voice quality is excellent.
Accompanying drawing explanation
Fig. 1 is fundamental diagram of the present utility model.
Fig. 2 is circuit system structure chart of the present utility model.
Fig. 3 is system power supply structure chart of the present utility model.
Fig. 4 is SIP information master drawing in server registration of the present utility model and session.
Fig. 5 is SIP log-on message master drawing of the present utility model.
Wherein, 101, 201, 301-FM1188 noise reduction chip, 102, 202, 302-CS42L52 speech chip, 103-G.711 speech coding module, 104-G.711 tone decoding module, 105-RTP package module, 106-RTP unpacks module, 107-UDP transport layer, 108-SIP/SDP protocol stack module, 210, 310-STM32F401 master chip, 211, 311-LCD1602(industry character liquid crystal module), 212-is with LED switch power supply, 213, 313-W5200 ethernet controller, 214-MIC (microphone), 215-SPEAKER (loud speaker), 216-RJ45 interface, 217-SPI (Serial Peripheral Interface (SPI)), 218-I2S (serial digital audio bus), 320-TPS 54294 PWPR Buck Regulator (switching pressurizer), 321-Adapter (power supply adaptor), 322-LD39060PU33R (filtering chip), (323-AMS1117_2.5 forward low pressure drop voltage stabilizing chip), 324-HT7218(lowering and stabilizing blood pressure chip), 219-GPIO(universal input/output interface).
Embodiment
In order to understand the utility model better, illustrate content of the present utility model further below in conjunction with drawings and Examples.
Operation principle as shown in Figure 1, IP phone transmission method described in the utility model, the collection of voice signal is started after comprising the registrar invitation checking of transmission both sides, the coding of audio signal, RTP voice data package, UDP exhalation datagram transmission, the response of protocol stack module and UDP transport layer, protocol stack module is to the control of audio collection, protocol stack module controls the selection of coder module, protocol stack module controls the selection of decoder module, the control that protocol stack module exports audio frequency, the response of protocol stack module and UDP transport layer, UDP incoming call datagram transmission, RTP voice data unpacks, the decoding of audio signal, the output of voice signal.
For realizing the transmission method of above-mentioned IP phone, concrete steps are as follows:
First, calling and called twocouese server sends Register(registration) message, server can return a checking message, after ethernet controller w5200 213,313 receives message, be sent to STM32F401RE master chip 210,310 to process, according to SIP part correlative code, extract wherein useful information, again send Register(registration) message is to server, if errorless, then succeed in registration, if wrong, need to resend once; To make server to be seen brassboard is registered.
During concrete enforcement, the PC selected holds sip server software miniSIPServer and X-Lite(VOIP soft phone freeware) as bitcom, the whole sip message in above-mentioned server registration and communication process are by packet catcher wireshark(network package analysis software) obtain (as shown in Figure 4).Such as: circuit board IP address will be set and be set to 192.168.0.4, the IP address of server and soft phone that number is set to 101, PC end is set to 192.168.0.212, and soft telephone number is set as 102; STM32F401RE master chip 210,310 Register(registration) message is to sip server, sip server can return a SIP Status(SIP state): 407 ProxyAuthentication Required(Proxy Authentications), mean and need by checking, server is by returning a piece of news as shown in Figure 5, i.e. Proxy-Authenticate:Digestrealm=" 192.168.0.212 ", Algorithm=MD5, nonce=" 45B454B
E405359A846DD2B57328A1437",stale=FALSE 。User need extract the message in nonce, then the user name of oneself, password, ip address and method MD5 are encrypted, form a new Proxy-Authenticate head, join new Register request, and then send out a Register request.Finally, after information is errorless, server can return 200 OK, namely succeeds in registration.
Secondly, it is SDP(Session Description Protocol that calling party sends out message body) invite(invite) converse to callee, described message body comprises institute's support voice coding/decoding type i P address, the information such as RTP port numbers; When information is errorless, calling party can receive 100Trying(on-state code 100) and 180Ringing(responsive state code 180), simultaneously RTP package can transmit " too " sound return, prompting caller is connected; When the other side agree to session and support invite(invite) in SDP(Session Description Protocol) comprise encoding and decoding speech time, calling party receives 200 OK(and terminates RINGING state code 200) after, reply an ACK(message acknowledge character) answer signal, call is started.
Then, when calling party speaks, voice signal is through MIC(microphone) 214, be sent to the Dolby circuit using FM1188 noise reduction chip 101,201,301, be sent to CS42L52 speech chip 102,202,302 again through Dolby circuit and form data to be encoded, be then sent to STM32F401RE master chip 210,310 by I2S (serial digital audio bus) 218 buses; In master chip, correlative code realizes using G.711 algorithm coding to voice signal, gives W5200 ethernet controller 213,313 afterwards, is packaged into RTP bag, is sent to object IP; When callee has transmitting voice signal to come, after W5200 ethernet controller 213, the 313 reception data of calling party, STM32F401RE master chip 210,310 correlative code carries out G.711 algorithm decoding, CS42L52 speech chip 102,202,302 is sent to by I2S (serial digital audio bus) 218, after processing, be sent to SPEAKER (loud speaker) 215 and play out sound..
Finally, when calling and called one side wants to terminate call, send Bye ending request, the other side returns a 200OK(and terminates RINGING state code 200) after, end of conversation.
For realizing above-mentioned IP telephone transport method, circuit system structure chart as shown in Figure 2, the transmission equipment of the IP phone built comprises W5200 Ethernet microcontroller 213,313, STM32F401 master chip 210,310, the noise reduction chip 201,301 of FM1188, CS42L52 speech chip 102,202,302, be positioned at the G.711 speech coding module 103 of STM32F401 master chip 210,310, G.711 tone decoding module 104, LCD1602(industry character liquid crystal module) 311, band LED switch power supply 212 etc.; And, the collection of audio frequency, export between same coding and decoding module and connected by I2S (serial digital audio bus) 218, STM32F401 master chip 210,310 is connected with W5200 Ethernet microcontroller 213,313 by SPI (Serial Peripheral Interface (SPI)) 217, and master chip is connected by general input/output interface GPIO and LCD1602 industry character liquid crystal module 211,311; W5200 Ethernet microcontroller 213, between 313 and Ethernet, carry out Signal transmissions by RJ45 interface 216.
Wherein, Switching Power Supply as shown in Figure 4, the power supply Adapter (power supply adaptor) 321 that equipment needs requires as the input of 12V, 0.5A direct current, TPS54294PWPR Buck Regulator(suitching type voltage-releasing voltage stabilizer) 320 output 3.3V direct voltage supply W5200 ethernet controllers 213,313, CS42L52 speech chip 102,202,302, FM1188 noise reduction chip 201,301; Export the LCD1602 industry character liquid crystal module 211,311 of 5V direct voltage supply LCDs, and respectively by exporting 3.3V direct voltage supply STM32F401 master chip 210,310 after LD39060PU33R (filtering chip) 322 filtering, export 2.5V direct voltage by AMS1117_2.5 (forward low pressure drop voltage stabilizing chip) 323 and supply CS42L52 speech chip 102,202,302 separately, by HT7218(lowering and stabilizing blood pressure chip) 324 provide 1.8V direct voltage to FM1188 noise reduction chip 201,301 separately.
It should be noted that the various equivalents that those skilled in the art do the utility model or amendment, all within claims limited range listed by the application.
Claims (4)
1. a signal transmission apparatus for IP phone, is characterized in that: comprise Ethernet microcontroller, noise reduction chip, speech chip, the master chip containing audio coding module and audio decoder module, LCDs, Switching Power Supply, step-down chip; And, audio collection is connected by data wire with the audio frequency coding and decoding module of output module with master chip, master chip is connected with ethernet controller by Peripheral Interface, is connected with speech chip by Audio Data Line, is connected with LCDs by general-purpose interface.
2. the signal transmission apparatus of IP phone as claimed in claim 1, it is characterized in that: described noise reduction chip is made up of FM1188 noise reduction chip (101,201,301), speech chip is made up of CS42L52 speech chip (102,202,302).
3. the signal transmission apparatus of IP phone as claimed in claim 1 or 2, is characterized in that: described audio frequency coding and decoding module adopts the audio coding module (103) of G.711 algorithm, audio decoder module (104) is formed.
4. the signal transmission apparatus of IP phone as claimed in claim 1 or 2, is characterized in that: described ethernet controller is made up of W5200 ethernet controller (213,313).
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201420868288.XU CN204350045U (en) | 2014-12-31 | 2014-12-31 | A kind of signal transmission apparatus of IP phone |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN201420868288.XU CN204350045U (en) | 2014-12-31 | 2014-12-31 | A kind of signal transmission apparatus of IP phone |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| CN204350045U true CN204350045U (en) | 2015-05-20 |
Family
ID=53233282
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN201420868288.XU Expired - Fee Related CN204350045U (en) | 2014-12-31 | 2014-12-31 | A kind of signal transmission apparatus of IP phone |
Country Status (1)
| Country | Link |
|---|---|
| CN (1) | CN204350045U (en) |
Cited By (2)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN105717496A (en) * | 2016-01-30 | 2016-06-29 | 湖北工业大学 | Realization method of FDA (Frequency Diverse Array) MIMO (Multiple-Input Multiple-Output) radar system based on matrix completion |
| CN107979484A (en) * | 2016-10-25 | 2018-05-01 | 北京佳讯飞鸿电气股份有限公司 | A kind of scheduling system and method with a variety of communication functions |
-
2014
- 2014-12-31 CN CN201420868288.XU patent/CN204350045U/en not_active Expired - Fee Related
Cited By (3)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN105717496A (en) * | 2016-01-30 | 2016-06-29 | 湖北工业大学 | Realization method of FDA (Frequency Diverse Array) MIMO (Multiple-Input Multiple-Output) radar system based on matrix completion |
| CN105717496B (en) * | 2016-01-30 | 2017-11-10 | 湖北工业大学 | A kind of implementation method of the frequency control battle array MIMO radar system based on matrix fill-in |
| CN107979484A (en) * | 2016-10-25 | 2018-05-01 | 北京佳讯飞鸿电气股份有限公司 | A kind of scheduling system and method with a variety of communication functions |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| WO2007021446A8 (en) | Associating a telephone call with a dialog based on a computer protocol such as sip | |
| WO2012106894A1 (en) | Method and device for transmitting media stream data in cloud computing system | |
| CN102547416B (en) | Method for partially loading media based on mobile phone and television | |
| CN104954724B (en) | A kind of video call switching method, Intelligent television terminal, mobile terminal and system | |
| WO2012079510A1 (en) | Mute indication method and device applied to video conferencing | |
| WO2011109972A1 (en) | Method and system for implementing multimedia conference | |
| CN102299962A (en) | Cloud-based Voice over Internet Protocol (VoIP) system, device and method | |
| WO2014114085A1 (en) | Thin client and communication method and device thereof | |
| CN204350045U (en) | A kind of signal transmission apparatus of IP phone | |
| CN107040458B (en) | Method and system for realizing intercommunication of video conference | |
| WO2011130987A1 (en) | Method for faxing based on voice over internet protocol and system thereof | |
| US20090327426A1 (en) | Remote call control and conferencing using paired devices | |
| WO2014040433A1 (en) | Implementation method and implementation system for ptt call based on voip technology | |
| CN201323575Y (en) | Bluetooth headset | |
| CN103684970B (en) | The transmission method of media data flow and thin terminal | |
| US20110235632A1 (en) | Method And Apparatus For Performing High-Quality Speech Communication Across Voice Over Internet Protocol (VoIP) Communications Networks | |
| US8077745B2 (en) | Techniques for unidirectional disabling of audio-video synchronization | |
| WO2018076376A1 (en) | Voice data transmission method, user device, and storage medium | |
| WO2017152566A1 (en) | Method for negotiating media coding/decoding, and terminal device | |
| TWI435589B (en) | Voip integrating system and method thereof | |
| WO2012126336A1 (en) | Method and system for providing conference call function for common terminal | |
| CN103067627B (en) | Multichannel conversation fast-switching method based on VoIP (Voice over Internet Phone) system | |
| CN113612759A (en) | High-performance high-concurrency intelligent broadcasting system based on SIP protocol and implementation method | |
| KR100986113B1 (en) | The media codec sharing method for multi-party call | |
| CN101998003B (en) | Call holding method for voice over internet protocol (VoIP) based on session initiation protocol (SIP) |
Legal Events
| Date | Code | Title | Description |
|---|---|---|---|
| C14 | Grant of patent or utility model | ||
| GR01 | Patent grant | ||
| CF01 | Termination of patent right due to non-payment of annual fee |
Granted publication date: 20150520 Termination date: 20171231 |
|
| CF01 | Termination of patent right due to non-payment of annual fee |