CN102231731A - System and method for dynamically adjusting voice coding when wired telephone and wireless telephone intercommunicate - Google Patents
System and method for dynamically adjusting voice coding when wired telephone and wireless telephone intercommunicate Download PDFInfo
- Publication number
- CN102231731A CN102231731A CN2011101649025A CN201110164902A CN102231731A CN 102231731 A CN102231731 A CN 102231731A CN 2011101649025 A CN2011101649025 A CN 2011101649025A CN 201110164902 A CN201110164902 A CN 201110164902A CN 102231731 A CN102231731 A CN 102231731A
- Authority
- CN
- China
- Prior art keywords
- module
- server
- coding
- wired
- telephone
- 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.)
- Pending
Links
Images
Landscapes
- Telephonic Communication Services (AREA)
Abstract
Provided are a system and a method for dynamically adjusting voice coding when a wired telephone and a wireless telephone intercommunicate. The system comprises a wired IP telephone, an IMS/NGN voice server, a trunk gateway, a wired common telephone and a wireless mobile phone. A dynamic selection coding module is added to the wired common telephone. The concrete interaction module comprises a DSP module, a call control module, a number analysis module and a signaling processing module. A user dials an assigned number through the IP telephone and the IP telephone dynamically selects voice coding to negotiate with the server based on the called number belongs to a wired common telephone or a wireless mobile phone so as to dynamically adjust the coding. According the invention, dynamically adjusting the voice coding is realized through identifying the called number; hence the cost of server coding transition caused by different voice coding of two ends is avoided, the load of the server is reduced and the maintenance cost of the service provider is reduced.
Description
[technical field]
The invention belongs to the IP communication technique field, speech coding dynamic debugging system and method thereof when specifically being meant a kind of wire telephone and radio apparatus intercommunication.
[background technology]
Telecom communication has now entered the full-service epoch, and it is trend of the times that traditional PSTN speech communication technology is replaced by the IP mechanics of communication.At the IP communication field, domestic telecom operators generally adopt the coding of G711A/U at present, be the tonequality needs on the one hand, also be that other data services are such as needs such as fax, Modem on the other hand, and in field of wireless communication because bandwidth problem adopts is the speech coding of AMR, but tonequality is but equally matched with G711a/u.One road G711A/U occupied bandwidth is 64k, and one road AMR-NB/WB occupied bandwidth only needs about the 10K.
Wired or wireless terminal all is to encode according to the acquiescence of oneself to carry out communication at present, if the coding at two ends is different, needs local side to be equipped with special coding converting apparatus and realizes speech communication, has increased the cost of operator.
[summary of the invention]
Speech coding dynamic debugging system when one of technical problem to be solved by this invention is a kind of wire telephone cheaply and radio apparatus conversation are provided.
Speech coding dynamic adjusting method when two of technical problem to be solved by this invention is a kind of wire telephone cheaply and radio apparatus conversation are provided.
The present invention one of solves the problems of the technologies described above by the following technical solutions:
Speech coding dynamic debugging system when a kind of wire telephone and radio apparatus intercommunication, comprise the wired IP phone machine that to call, IMS/NGN voice server with CCF, can carry out the Tandem Gateway that code conversion is used, the wired ordinary telephone set that can answer, the wireless phone that can answer; Increase a Dynamic Selection coding module end in described wired IP phone machine, its concrete interactive module comprises: comprise the DSP module, Call Control Block, number analysis module, signaling processing module; Described DSP module: be responsible for detecting number and encoding and decoding; Described Call Control Block: be responsible for Business Processing and route; Described number analysis module: provide the called phone type according to number information; Described signaling processing module: the coding different according to the called phone type selecting.
Further, described wired IP phone machine is meant by being registered to and is connected to the sip terminal phone that described IMS/NGN sip server connects, and meets RFC3261 standard and related expanding standard.
Further, described wired IP phone machine is a common phone, is meant by the IAD/AG access device to be registered on the IMS/NGN sip server; Described IAD/AG meets RFC3261 standard and related expanding standard; Described IAD is meant integrated access equipment; Described AG is meant IAD.
Further, described IMS/NGN sip server is followed the sip server behavior of stipulating in RFC3261 standard and the related expanding standard.
Further, described Tandem Gateway is the server that is used for doing medium or signaling conversion in the IMS/NGN environment specially, and follows RFC3261 standard and related expanding standard.
Further, described wired ordinary telephone set is the phone that is connected on the PSTN network on the stored-program control exchange simulation mouth.Further, described radio apparatus comprises the wireless telecommunications standard that GSM, CDMA, TD-SCDMA, WCDMA, CDMA2000 meet GB.
The present invention solve the problems of the technologies described above by the following technical solutions two:
Speech coding dynamic adjusting method when a kind of wire telephone and radio apparatus intercommunication comprises the steps:
Step 10: the user uses wired IP phone machine off-hook to begin dialing;
Step 20:DSP module is collected number, and this number is sent to Call Control Block;
Step 30: Call Control Block is finished professional control, and this number is delivered to the number analysis module;
Step 40: the number analysis module is checked numbers and is analyzed or carry out number analysis according to predetermined number figure configuration according to the number database of operator oneself, matches corresponding clauses and subclauses;
Step 50: judge that the type of number is the wireless phone number? be, then change step 60 over to, not, then change step 70 over to;
Step 60: the number analysis module is delivered to signaling processing module with the conclusion of analyzing and is confirmed to adopt the AMR coding;
Step 61: signaling processing module sends SIP signaling Invite message and carries out the AMR conversation to the IMS/NGN server requirement;
Signaling message is replied in the unified request of step 62:IMS/NGN server, and signaling processing module receives after the confirmation message that notice DSP module starts the conversation of AMR coding; Change step 80 over to;
Step 70: the number analysis module is delivered to signaling processing module with the conclusion of analyzing and is confirmed to adopt the G711A/U coding;
Step 71: signaling processing module sends SIP signaling Invite message and carries out the G711A/U conversation to the IMS/NGN server requirement;
Signaling message is replied in the unified request of step 72:IMS/NGN server, and signaling processing module receives after the confirmation message that notice DSP module starts the conversation of G711A/U coding;
Step 80: finish.
The invention has the advantages that: realize dynamically adjusting speech coding by identification, thereby avoided the different expense that causes the server code conversion of two ends speech coding, alleviate load of server, thereby reduce the maintenance cost of operator according to called number.
[description of drawings]
The invention will be further described in conjunction with the embodiments with reference to the accompanying drawings.
Fig. 1 is Dynamic Selection coding module figure among the present invention.
Fig. 2 is the first embodiment of the invention workflow diagram.
Fig. 3 is a number figure allocation plan among the present invention.
Fig. 4 is the second embodiment of the invention workflow diagram.
[embodiment]
Speech coding dynamic debugging system when a kind of wire telephone and radio apparatus intercommunication, comprise the wired IP phone machine that to call, IMS/NGN voice server with CCF, can carry out the Tandem Gateway that code conversion is used, the wired ordinary telephone set that can answer, the wireless phone that can answer.
Increase a Dynamic Selection coding module end in wired IP phone machine, as shown in Figure 1, its concrete interactive module comprises: comprise DSP (Digit signal processor) module, Call Control Block, number analysis module, signaling processing module; The DSP module is responsible for detecting number and encoding and decoding; Call Control Block is responsible for Business Processing and route; The number analysis module provides the called phone type according to number information; The coding that signaling processing module is different according to the called phone type selecting, the signaling here adopts SIP (SessionInitiation Protocol).
Wired IP phone machine is meant by being registered to and is connected to the sip terminal phone that described IMS/NGN sip server connects, and meets RFC3261 standard and related expanding standard.Wired IP phone machine can be a common phone also, is meant by the IAD/AG access device to be registered on the IMS/NGN sip server; Described IAD/AG meets RFC3261 standard and related expanding standard; Described IAD is meant integrated access equipment; Described AG is meant IAD.Described IMS/NGN sip server is followed the sip server behavior of stipulating in RFC3261 standard and the related expanding standard.Described Tandem Gateway is the server that is used for doing medium or signaling conversion in the IMS/NGN environment specially, and follows RFC3261 standard and related expanding standard.
Wired ordinary telephone set is the phone that is connected on the PSTN network on the stored-program control exchange simulation mouth.Radio apparatus comprises the wireless telecommunications standard that GSM, CDMA, TD-SCDMA, WCDMA, CDMA2000 etc. meet GB.
Below by two embodiment explanations concrete workflow of the present invention when called number is respectively radio apparatus and fixed line.
First embodiment:
See also shown in Figure 2ly, the speech coding dynamic adjusting method the during conversation of wired IP phone machine and wireless phone comprises the steps:
Step 10: the user uses wired IP phone machine off-hook to begin dialing, and number is 13900001234;
Step 20:DSP module is collected this number, and this number is sent to Call Control Block;
Step 30: Call Control Block is finished professional control, and this number is delivered to the number analysis module;
Step 40: the number analysis module is checked numbers and is carried out number analysis according to number figure configuration as shown in Figure 3, has matched 1[358] these clauses and subclauses of XXXXXXXXX;
Step 50: confirm that the type of number is the wireless phone number;
Step 60: the number analysis module is delivered to signaling processing module with the conclusion of analyzing and is confirmed to adopt the AMR coding;
Step 61: signaling processing module sends SIP (Session Initiation Protocol) signaling Invite message and carries out the AMR conversation to the IMS/NGN server requirement;
Signaling processing module sends the message of Invite according to the number analysis result, and wherein SDP (SessionDescription Protocol) comprises the statement of AMR, and the message parameter format is as follows:
INVITE?sip：13900001234ims.fj.chinamobile.com?SIP/2.0
Call-ID：033ea9885ec3cf00ims.fj.chinamobile.com
Via：SIP/2.0/UDP?192.168.180.132:5060；branch＝z9hG4bK3439c6817742680
Max-Forwards：70
Supported：100rel，replaces
Contact：<sip:+8658128053333192.168.180.132:5060>
From：
″+8658128053333″<sip:+8658128053333ims.fj.chinamobile.com>；tag＝5462f5
21
To：<sip:13900001234ims.fj.chinamobile.com>
CSeq：1INVITE
Content-Type：application/sdp
Content-Length：139
v＝0
o＝sipua?396236?396236?IN?IP4?192.168.180.132
s＝IADCall
c＝IN?IP4?192.168.180.132
t＝0?0
m＝audio?10048?RTP/AVP?119?120
a＝rtpmap：119?AMR/8000
a＝rtpmap：120?AMR-WB/16000
a＝sendrecv
Wherein 119,120 of the SDP the inside indicate to use the sample rate of AMR to converse as the coding of 8K or 16K;
Step 62:IMS/NGN server is forwarded to Tandem Gateway application resource with message after receiving message, after the Tandem Gateway resource bid is arrived, and notice IMS/NGN server, server is responded SIP signaling 200OK to ip voice equipment.The notification call control module started the AMR voice call after signaling processing module was received confirmation message; Change step 80 over to;
Message format is as follows:
SIP/2.0?200?OK
Call-ID：YzI0ZDNjNzQxZTE5ZDM5OThkMDI4MjIxY2UzMTBjMTg.
Via： SIP/2.0/UDP
192.168.180.127:5060；branch＝z9hG4bKrhdq2m007oi0je8cV7o1.1
Supported：100rel，replaces
Contact：<sip:13900001234192.168.180.132:5060>
From：
″+8658128053333″<sip:+8658128053333ims.fj.chinamobile.com>；tag＝33ca58
79eb.rzsorwyosuworxw-vqwq
To：<sip:13900001234ims.fj.chinamobile.com>；tag＝48bee4b8
CSeq：1?INVITE
Content-Type：application/sdp
Content-Length：139
v＝0
o＝sipua?396236?396236?IN?IP4?192.168.180.133
s＝IADCall
c＝IN?IP4?192.168.180.132
t＝0?0
m＝audio?10048?RTP/AVP?119?120
a＝rtpmap:119AMR/8000
a＝rtpmap:120AMR-WB/16000
a＝sendrecv
Step 80: finish.So far complete voice call is set up.
Second embodiment:
If what the user dialed is fixed line number (059528053888), see also shown in Figure 4ly, the speech coding dynamic adjusting method the during conversation of wired IP phone machine and fixed line comprises the steps:
Step 10: the user uses wired IP phone machine off-hook to begin dialing, and number is 059528053888;
Step 20:DSP module is collected this number, and this number is sent to Call Control Block;
Step 30: Call Control Block is finished professional control, and this number is delivered to the number analysis module;
Step 40: the number analysis module is checked numbers and is carried out number analysiss according to number figure configuration as shown in Figure 3;
Step 50: the number analysis module owing to do not match, is therefore thought fixed line in number figure configuration; Change step 70 over to;
Step 70: the number analysis module is delivered to the coding that signaling processing module confirms to adopt G711A/U with the conclusion of analyzing;
Step 71: signaling processing module sends SIP (Session Initiation Protocol) signaling Invite message and carries out the G711A/U conversation to the IMS/NGN server requirement;
Signaling processing module sends the message of Invite according to the number analysis result, and the message parameter format is as follows:
INVITE?sip：059528053888ims.fj.chinamobile.com?SIP/2.0
Call-ID：033ea9885ec3cf00ims.fj.chinamobile.com
Via：SIP/2.0/UDP?192.168.180.132:5060；branch＝z9hG4bK3439c6817742680
Max-Forwards：70
Supported：100rel，replaces
Contact：<sip:+8658128053333192.168.180.132:5060>
From：
″+8658128053333″<sip:+8658128053333ims.fj.chinamobile.com>；tag＝5462f5
21
To：<sip:059528053888ims.fj.chinamobile.com>
CSeq：1?INVITE
Content-Type：application/sdp
Content-Length：139
v＝0
o＝sipua?396236?396236?IN?IP4?192.168.180.132
s＝IADCall
c＝IN?IP4?192.168.180.132
t＝0?0
m＝audio?10048?RTP/AVP?0?8
a＝rtpmap:0?PCMU/8000
a＝rtpmap:8?PCMA/8000
a＝sendrecv
Wherein PCMU/PCMA is exactly the speech coding of G711A/U;
Step 72:IMS/NGN server is forwarded to Tandem Gateway application resource with message after receiving message, after the Tandem Gateway resource bid is arrived, and notice IMS/NGN server, server is responded SIP signaling 200OK to ip voice equipment.The notification call control module started the G711A/U voice call after signaling processing module was received confirmation message; Change step 80 over to;
Message format is as follows:
SIP/2.0200?OK
Call-ID：YzI0ZDNjNzQxZTE5ZDM5OThkMDI4MjIxY2UzMTBjMTg.
Via： SIP/2.0/UDP
192.168.180.127:5060；branch＝z9hG4bKrhdq2m007oi0je8cv7o1.1
Supported：100rel，replaces
Contact：<sip:059528053888192.168.180.132:5060>
From：
″+8658128053333″<sip:+8658128053333ims.fj.chinamobile.com>；tag＝33ca58
79eb.rzsorwyosuworxw-vqwq
To：<sip:059528053888ims.fj.chinamobile.com>；tag＝48bee4b8
CSeq：1INVITE
Content-Type：application/sdp
Content-Length：139
v＝0
o＝sipua?396236?396236IN?IP4?192.168.180.133
s＝IADCall
c＝IN?IP4?192.168.180.132
t＝0?0
m＝audio?10048?RTP/AVP?0?8
a＝rtpmap:0?PCMU/8000
a＝rtpmap：8?PCMA/8000
a＝sendrecv；
Step 80: finish.So far complete voice call is set up.
As seen, by technical scheme of the present invention, IMS/NGN does not need to do any code conversion, and the ip voice phone carries out coding negotiation according to the automatic identification number type of number rule with local side, thereby realizes the ability of adaptive coding.
The above is preferred embodiment of the present invention only, is not to be used to limit protection model figure of the present invention.Within the spirit and principles in the present invention all, any modification of being done, be equal to and replace and improvement etc., all should be included within protection scope of the present invention.
Claims (8)
1. the speech coding dynamic debugging system when wire telephone and radio apparatus intercommunication, comprise the wired IP phone machine that to call, IMS/NGN voice server with CCF, can carry out the Tandem Gateway that code conversion is used, the wired ordinary telephone set that can answer, the wireless phone that can answer;
It is characterized in that: increase a Dynamic Selection coding module end in described wired IP phone machine, its concrete interactive module comprises: comprise the DSP module, Call Control Block, number analysis module, signaling processing module;
Described DSP module: be responsible for detecting number and encoding and decoding;
Described Call Control Block: be responsible for Business Processing and route;
Described number analysis module: provide the called phone type according to number information;
Described signaling processing module: the coding different according to the called phone type selecting.
2. the speech coding dynamic debugging system when wire telephone as claimed in claim 1 and radio apparatus intercommunication, it is characterized in that: described wired IP phone machine is meant by being registered to and is connected to the sip terminal phone that described IMS/NGN sip server connects, and meets RFC3261 standard and related expanding standard.
3. the speech coding dynamic debugging system when wire telephone as claimed in claim 1 and radio apparatus intercommunication is characterized in that: described wired IP phone machine is a common phone, is meant by the IAD/AG access device to be registered on the IMS/NGN sip server; Described IAD/AG meets RFC3261 standard and related expanding standard.
4. the speech coding dynamic debugging system when wire telephone as claimed in claim 1 and radio apparatus intercommunication is characterized in that: described IMS/NGN sip server is followed the sip server behavior of stipulating in RFC3261 standard and the related expanding standard.
5. the speech coding dynamic debugging system when wire telephone as claimed in claim 1 and radio apparatus intercommunication, it is characterized in that: described Tandem Gateway is the server that is used for doing medium or signaling conversion in the IMS/NGN environment specially, and follows RFC3261 standard and related expanding standard.
6. the speech coding dynamic debugging system when wire telephone as claimed in claim 1 and radio apparatus intercommunication is characterized in that: described wired ordinary telephone set is the phone that is connected on the PSTN network on the stored-program control exchange simulation mouth.
7. the speech coding dynamic debugging system when wire telephone as claimed in claim 1 and radio apparatus intercommunication is characterized in that: described radio apparatus comprises the wireless telecommunications standard that GSM, CDMA, TD-SCDMA, WCDMA, CDMA2000 meet GB.
8. the speech coding dynamic adjusting method when wire telephone and radio apparatus intercommunication is characterized in that: comprise the steps:
Step 10: the user uses wired IP phone machine off-hook to begin dialing;
Step 20:DSP module is collected number, and this number is sent to Call Control Block;
Step 30: Call Control Block is finished professional control, and this number is delivered to the number analysis module;
Step 40: the number analysis module is checked numbers and is analyzed or carry out number analysis according to predetermined number figure configuration according to the number database of operator oneself, matches corresponding clauses and subclauses;
Step 50: judge that the type of number is the wireless phone number? be, then change step 60 over to, not, then change step 70 over to;
Step 60: the number analysis module is delivered to signaling processing module with the conclusion of analyzing and is confirmed to adopt the AMR coding;
Step 61: signaling processing module sends SIP signaling Invite message and carries out the AMR conversation to the IMS/NGN server requirement;
Signaling message is replied in the unified request of step 62:IMS/NGN server, and signaling processing module receives after the confirmation message that notice DSP module starts the conversation of AMR coding; Change step 80 over to;
Step 70: the number analysis module is delivered to signaling processing module with the conclusion of analyzing and is confirmed to adopt the G711A/U coding;
Step 71: signaling processing module sends SIP signaling Invite message and carries out the G711A/U conversation to the IMS/NGN server requirement;
Signaling message is replied in the unified request of step 72:IMS/NGN server, and signaling processing module receives after the confirmation message that notice DSP module starts the conversation of G711A/U coding;
Step 80: finish.
Priority Applications (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN2011101649025A CN102231731A (en) | 2011-06-17 | 2011-06-17 | System and method for dynamically adjusting voice coding when wired telephone and wireless telephone intercommunicate |
Applications Claiming Priority (1)
| Application Number | Priority Date | Filing Date | Title |
|---|---|---|---|
| CN2011101649025A CN102231731A (en) | 2011-06-17 | 2011-06-17 | System and method for dynamically adjusting voice coding when wired telephone and wireless telephone intercommunicate |
Publications (1)
| Publication Number | Publication Date |
|---|---|
| CN102231731A true CN102231731A (en) | 2011-11-02 |
Family
ID=44844264
Family Applications (1)
| Application Number | Title | Priority Date | Filing Date |
|---|---|---|---|
| CN2011101649025A Pending CN102231731A (en) | 2011-06-17 | 2011-06-17 | System and method for dynamically adjusting voice coding when wired telephone and wireless telephone intercommunicate |
Country Status (1)
| Country | Link |
|---|---|
| CN (1) | CN102231731A (en) |
Citations (6)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US20030193946A1 (en) * | 1998-12-17 | 2003-10-16 | Symbol Technologies, Inc., A Delaware Corporation | Apparatus for interfacing a wireless local network and a wired voice telecommunications system |
| CN1471296A (en) * | 2002-07-24 | 2004-01-28 | Lg������ʽ���� | System and method for operating terminal in at least two communication modes |
| CN1784026A (en) * | 2004-11-30 | 2006-06-07 | 中国科学院声学研究所 | Speech communication system and method based on mobile telephone speech encoding and decoding system |
| CN1842066A (en) * | 2005-03-29 | 2006-10-04 | 鸿富锦精密工业（深圳）有限公司 | Network communication equipment and transmission system using same |
| WO2008021547A2 (en) * | 2006-08-18 | 2008-02-21 | At & T Mobility Ii Llc | Dual mode service wifi access control |
| CN201252627Y (en) * | 2008-09-04 | 2009-06-03 | 深圳市泰丰网络设备有限公司 | Multi-module communication terminal |
-
2011
- 2011-06-17 CN CN2011101649025A patent/CN102231731A/en active Pending
Patent Citations (6)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| US20030193946A1 (en) * | 1998-12-17 | 2003-10-16 | Symbol Technologies, Inc., A Delaware Corporation | Apparatus for interfacing a wireless local network and a wired voice telecommunications system |
| CN1471296A (en) * | 2002-07-24 | 2004-01-28 | Lg������ʽ���� | System and method for operating terminal in at least two communication modes |
| CN1784026A (en) * | 2004-11-30 | 2006-06-07 | 中国科学院声学研究所 | Speech communication system and method based on mobile telephone speech encoding and decoding system |
| CN1842066A (en) * | 2005-03-29 | 2006-10-04 | 鸿富锦精密工业（深圳）有限公司 | Network communication equipment and transmission system using same |
| WO2008021547A2 (en) * | 2006-08-18 | 2008-02-21 | At & T Mobility Ii Llc | Dual mode service wifi access control |
| CN201252627Y (en) * | 2008-09-04 | 2009-06-03 | 深圳市泰丰网络设备有限公司 | Multi-module communication terminal |
Similar Documents
| Publication | Publication Date | Title |
|---|---|---|
| CN1659921B (en) | Signaling method for wireless network | |
| US20090003570A1 (en) | Method, system and apparatus for providing endpoint-to-endpoint transcoding free connection | |
| CN1941783B (en) | Intelligent border element | |
| EP2088735A1 (en) | Client side media splitting function | |
| CN100466761C (en) | Mobile switching centre built in video gateway and method for realizing multimedia interconnection | |
| US8457139B2 (en) | Systems, methods, and computer readable media for loose coupling of multimedia resource function controllers and multimedia resource function processors | |
| WO2012063889A1 (en) | Core network and communication system | |
| US20130223304A1 (en) | Core network and communication system | |
| CN101094086B (en) | Method and system for constructing call canter by next generation of network | |
| CN102843336A (en) | Method and system for accessing IMS (IP Multimedia Subsystem) multimedia conference | |
| CN101005402B (en) | Information report method of SIP user agent service switching | |
| CN201893844U (en) | Call center system based on internet protocol-private branch exchange (IP-PBX) | |
| CN100417245C (en) | PTT service realizing system and method based on VoIP technique | |
| US20050018652A1 (en) | System and method for proxy gatekeeper in H.323 based IP telephony systems | |
| US8355323B2 (en) | Interactive voice response and secure media system integration | |
| CN102271137A (en) | Media server | |
| CN101860630B (en) | Transmission method and system of out-of-band dual tone multiple frequency (DTMF) signal | |
| US20130231103A1 (en) | Core network and communication system | |
| CN101273609A (en) | Telephone call processing method and apparatus | |
| CN100571307C (en) | In next generation network, insert the method for integrated services digital network console | |
| CN101325630A (en) | Network telephone system and operation method thereof | |
| CN202111736U (en) | Voice coding dynamic adjustment system for intercommunication between wired phone and wireless phone | |
| CN105306420B (en) | Realize the method, apparatus played from Text To Speech cycle of business operations and server | |
| CN102231731A (en) | System and method for dynamically adjusting voice coding when wired telephone and wireless telephone intercommunicate | |
| CN107613123A (en) | The system and method for wireless telephone function is realized based on WIFI |
Legal Events
| Date | Code | Title | Description |
|---|---|---|---|
| C06 | Publication | ||
| PB01 | Publication | ||
| C10 | Entry into substantive examination | ||
| SE01 | Entry into force of request for substantive examination | ||
| C02 | Deemed withdrawal of patent application after publication (patent law 2001) | ||
| WD01 | Invention patent application deemed withdrawn after publication |
Application publication date: 20111102 |