CN101997997A - System for realizing Internet telephony call transfer by utilizing SIP protocol and method thereof - Google Patents
System for realizing Internet telephony call transfer by utilizing SIP protocol and method thereof Download PDFInfo
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- CN101997997A CN101997997A CN2010105498621A CN201010549862A CN101997997A CN 101997997 A CN101997997 A CN 101997997A CN 2010105498621 A CN2010105498621 A CN 2010105498621A CN 201010549862 A CN201010549862 A CN 201010549862A CN 101997997 A CN101997997 A CN 101997997A
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Abstract
The invention discloses a system for realizing Internet telephony call transfer by utilizing SIP protocol, comprising Internet phoney terminals, SIP gateways and soft switch equipment; wherein each SIP gateway is connected with one Internet phoney terminal, the soft switch equipment is connected with the SIP gateway by virtue of INTERNET network, and the soft switch equipment and a plurality of SIP gateways form a one-to-many internetworking system. Simple INFO method is adopted to realize complex call service, and call process is tightly combined with the soft switch equipment, thus the method is simple and efficient.
Description
Technical field
The invention belongs to the Internet telephony field, be specifically related to a kind of Session Initiation Protocol that utilizes and realize the system and method that the Internet telephone calls shifts.
Background technology
In recent years, Internet has obtained develop rapidly and popularization and application, and has obtained extensive approval as the dominant position of IP protocol architecture in the data network framework of its core technology.Simultaneously, along with the development of aggregation networks research in the IP technological frame and the proposition of voip technology, data network communications has incorporated traditional voice service field system.
The traditional PSTN phone is exactly its price, particularly toll telephone of standding high above the masses for This is what people generally disapprove of always, because cable is with high costs, so the charging way of phone is speech range and expense positive correlation.Compare with traditional PSTN net phone, the most conspicuous advantage of VoIP just is the price that it is cheap.And the reason of bringing this result is that the data transmission of VoIP is different with traditional phone, and it is not by special speech network, but by now extensively having covered the Internet in the world
Broadband, the IP of network changes into the inexorable trend into whole telecommunications network development.In visible future, IP phone will progressively replace black phone and final IPization fully.Develop and improve voip technology and then become a current research focus with comprehensive replacement PSTN.The VOIP agreement that exists has MGCP at present, H.323, H.248, SIP, wherein Session Initiation Protocol is the specified protocol of communication of future generation.
Summary of the invention
The object of the invention is to provide a kind of Session Initiation Protocol that utilizes to realize the system that the Internet telephone calls shifts, and is the ways and means that a kind of new the Internet telephone calls shifts, and has characteristics such as implementation method is simple, efficient.
In order to solve these problems of the prior art, technical scheme provided by the invention is:
A kind of Session Initiation Protocol that utilizes is realized the system that the Internet telephone calls shifts, comprise network telephone terminal, SIP gateway and Softswitch, it is characterized in that each SIP gateway connects 1 network telephone terminal, described Softswitch is connected by the INTERNET network with the SIP gateway, and the network interconnection system of Softswitch and several SIP gateways formation one-to-many.
The present invention also provides a kind of Session Initiation Protocol that utilizes to realize the method that the Internet telephone calls shifts, and it is characterized in that said method comprising the steps of:
(1) calling party sets up to converse by SIP gateway and Softswitch with first callee and is connected, and passes on the call transfer request to first callee, and first callee carries out call transfer by SIP gateway and Softswitch communication request;
(2) Softswitch sends to the calling party by the SIP gateway and sends first callee request of dialling that requires by the SIP gateway to first callee after conversation keeps request;
Transmit second callee's number by SIP gateway and Softswitch communication after (3) first callees dial, Softswitch makes the calling party and second callee set up conversation by the SIP gateway.
Preferably, carry out transmission parameter or instruction by INFO between SIP gateway and Softswitch in the described method.
Preferably, the SIP gateway is provided with the monitor of the command information of monitoring network telephone terminal in the described method, and described monitor detects dialing command information, the FLASH command information of network telephone terminal.
Preferably, after the calling party dials in the described method, the SIP gateway detects the dialing instruction, and dialed number sent to Softswitch by invite message, after described Softswitch receives invite information, send invite and ask the appointment callee of dialed number, specify callee's phone shake bell to remind; When specifying callee's off-hook, calling party and appointment callee set up conversation.
Preferably, in the described method when first callee presses phone FLASH key, the SIP gateway monitors the instruction message of FLASH key and the FLASH command information is transferred to Softswitch by INFO, described Softswitch sends re-invite message by the SIP gateway to the calling party, and conversation is kept.
Preferably, after described method step (3) Softswitch receives second callee's number information, Softswitch is initiated new calling to second callee, obtain second callee's media parameter, use re-invite message modification calling party's media parameter simultaneously, make the calling party and second callee set up conversation.
Principle of the present invention is based on the INFO method of RFC2976 regulation, the INFO method is used for transmitting the calling signal message along the session signal path in this regulation, be not to be used to change the state that SIP calls out, neither be used for changing by SIP initialization ground session status.Yet, can further strengthen the function of application of SIP in the technical solution of the present invention by the option information that increases, utilize the INFO method, can realize call flow such as calling transfer that some are special, realizes more talk business functions of assisting flexibly.Use the implementation of INFO method to need in the technical solution of the present invention and soft switch closely cooperates, use INFO to transmit DTMF and hooking action, can realize business such as calling transfer and Call Waiting, wherein the method for calling transfer realization is as follows:
At first user A calling party B sets up session.The process of calling out is initiated by invite according to the Session Initiation Protocol standard, and ACK confirms to set up conversation.
User A request user B arrives user C with call transfer, user B presses phone flash key, after the SIP gateway detects the FLSH key, give soft switch by INFO with this event report, after soft switch receives INFO, take out the content of INFO, discovery is the FLASH incident, soft switch sends re-invite message user A is called out maintenance, then, soft switch sends INFO to user B, the INFO body has the request that requirement user B plays dialing tone, after gateway is received INFO, check message body, play dialing tone and give user B, user B dials the number of user C at this moment, gateway sends to soft switch with the number that user B dials with INFO, after soft switch receives INFO, takes out the content in the INFO body, discovery is a string telephone number, soft switch is initiated new calling to this telephone number, obtains the media parameter of C, uses the media parameter of re-invite message modification user A simultaneously, make user A and user C set up conversation, user B this moment can on-hook, like this, uses the INFO method to realize the function of calling transfer.Technical scheme makes full use of the existing INFO of SIP like this, and combines closely with soft switch, realizes auxiliary activities functions such as some calling transfer, Call Waiting.
With respect to scheme of the prior art, advantage of the present invention is:
Technical solution of the present invention realizes the special calling flow process based on the SIP extended method, some complicated call business particularly, utilize existing simple INFO method, transmit some signals in the communication process, realize complicated call business, call flow and Softswitch are combined closely, and method is simply efficient.
Description of drawings
Below in conjunction with drawings and Examples the present invention is further described:
Fig. 1 utilizes Session Initiation Protocol to realize the system architecture diagram that the Internet telephone calls shifts;
Fig. 2 utilizes Session Initiation Protocol to realize the method flow diagram that the Internet telephone calls shifts.
Embodiment
Below in conjunction with specific embodiment such scheme is described further.Should be understood that these embodiment are used to the present invention is described and are not limited to limit the scope of the invention.The implementation condition that adopts among the embodiment can be done further adjustment according to the condition of concrete producer, and not marked implementation condition is generally the condition in the normal experiment.
Embodiment
As shown in Figure 1, present embodiment is realized network phone system framework such as Fig. 1 of following Session Initiation Protocol of calling transfer, comprise network telephone terminal, SIP gateway and Softswitch, each SIP gateway connects 1 network telephone terminal, described Softswitch is connected by the INTERNET network with the SIP gateway, and the network interconnection system of Softswitch and several SIP gateways formation one-to-many.
When carrying out calling transfer, the calling party sets up conversation with first callee by SIP gateway and Softswitch and is connected, and passes on the call transfer request to first callee, and first callee carries out call transfer by SIP gateway and Softswitch communication request; Softswitch sends conversation by the SIP gateway to the calling party and keeps the request back to send first callee request of dialling that requires by the SIP gateway to first callee; Transmit second callee's number by SIP gateway and Softswitch communication after first callee dials, Softswitch makes the calling party and second callee set up conversation by the SIP gateway.
Wherein carry out transmission parameter or instruction by INFO between SIP gateway and Softswitch.The SIP gateway is provided with the monitor of the command information of monitoring network telephone terminal, and described monitor detects dialing command information, the FLASH command information of network telephone terminal.After calling party's dialing, the SIP gateway detects the dialing instruction, and dialed number is sent to Softswitch by invite message, after described Softswitch receives invite information, send invite and ask the appointment callee of dialed number, specify callee's phone shake bell to remind; When specifying callee's off-hook, calling party and appointment callee set up conversation.When first callee presses phone FLASH key, the SIP gateway monitors the instruction message of FLASH key and the FLASH command information is transferred to Softswitch by INFO, described Softswitch sends re-invite message by the SIP gateway to the calling party, and conversation is kept.After Softswitch receives second callee's number information, Softswitch is initiated new calling to second callee, obtain second callee's media parameter, use re-invite message modification calling party's media parameter simultaneously, make the calling party and second callee set up conversation.
As shown in Figure 2, be the process mode figure that A, B, C realize calling transfer.A is the calling party, and B is first callee, and C is second callee, by system as shown in Figure 1, and utilizes INFO to do some expansions, transmits some signals in conversation, realizes complicated calling transfer.
User A calling party B at first, user A off-hook, hear dialing tone after, dial the number of user B, after gateway detects dialing, send invite message to soft switch, after soft switch receives invite message, send invite to user B, the gateway of user B shakes bell to phone, behind the user B off-hook, conversation is set up, and user A and user B converse.
User A request user B arrives user C with call transfer, user B presses phone flash key, after the SIP gateway detects the FLSH key, give soft switch by INFO with this event report, after soft switch receives INFO, take out the content of INFO, discovery is the FLASH incident, soft switch sends re-invite message user A is called out maintenance, then, soft switch sends INFO to user B, the INFO body has the request that requirement user B plays dialing tone, after gateway is received INFO, check message body, play dialing tone and give user B, user B dials the number of user C at this moment, gateway sends to soft switch with the number that user B dials with INFO, after soft switch receives INFO, takes out the content in the INFO body, discovery is a string telephone number, soft switch is initiated new calling to this telephone number, obtains the media parameter of C, uses the media parameter of re-invite message modification user A simultaneously, make user A and user C set up conversation, user B this moment can on-hook, like this, uses the INFO method to realize the function of calling transfer.
The flow process of calling out is as shown below:
The INFO method form of expansion is as follows:
INFO?sip:alice@pc33.example.com?SIP/2.0
Via：SIP/2.0/UDP?192.0.2.2:5060；branch＝z9hG4bKnabcdef
To：Bob<sip:bob@example.com>；tag＝a6c85cf
From：Alice<sip:alice@example.com>；tag＝1928301774
Call-Id：a84b4c76e66710@pc33.example.com
CSeq：314333?INFO
Content-type:application/extension
Content-Disposition:Info-Package
Content-length:24
Signal＝dial?tone；
Utilize existing simple INFO method, after above-mentioned INFO expansion, can transmit some signals in the communication process like this, realize complicated call business such as calling transfer or Call Waiting, call flow and Softswitch have been realized combining closely.
Above-mentioned example only is explanation technical conceive of the present invention and characteristics, and its purpose is to allow the people who is familiar with this technology can understand content of the present invention and enforcement according to this, can not limit protection scope of the present invention with this.All equivalent transformations that spirit is done according to the present invention or modification all should be encompassed within protection scope of the present invention.
Claims (7)
1. one kind is utilized Session Initiation Protocol to realize the system that the Internet telephone calls shifts, comprise network telephone terminal, SIP gateway and Softswitch, it is characterized in that each SIP gateway connects 1 network telephone terminal, described Softswitch is connected by the INTERNET network with the SIP gateway, and the network interconnection system of Softswitch and several SIP gateways formation one-to-many.
2. one kind is utilized Session Initiation Protocol to realize the method that the Internet telephone calls shifts, and it is characterized in that said method comprising the steps of:
(1) calling party sets up to converse by SIP gateway and Softswitch with first callee and is connected, and passes on the call transfer request to first callee, and first callee carries out call transfer by SIP gateway and Softswitch communication request;
(2) Softswitch sends to the calling party by the SIP gateway and sends first callee request of dialling that requires by the SIP gateway to first callee after conversation keeps request;
Transmit second callee's number by SIP gateway and Softswitch communication after (3) first callees dial, Softswitch makes the calling party and second callee set up conversation by the SIP gateway.
3. method according to claim 2 is characterized in that carrying out transmission parameter or instruction by INFO between SIP gateway and Softswitch in the described method.
4. method according to claim 2 is characterized in that SIP gateway in the described method is provided with the monitor of the command information of monitoring network telephone terminal, and described monitor detects dialing command information, the FLASH command information of network telephone terminal.
5. method according to claim 2, after it is characterized in that the calling party dials in the described method, the SIP gateway detects the dialing instruction, and dialed number sent to Softswitch by invite message, after described Softswitch receives invite information, send invite and ask the appointment callee of dialed number, specify callee's phone shake bell to remind; When specifying callee's off-hook, calling party and appointment callee set up conversation.
6. method according to claim 2, it is characterized in that in the described method when first callee presses phone FLASH key, the SIP gateway monitors the instruction message of FLASH key and the FLASH command information is transferred to Softswitch by INFO, described Softswitch sends re-invite message by the SIP gateway to the calling party, and conversation is kept.
7. method according to claim 2, after it is characterized in that described method step (3) Softswitch receives second callee's number information, Softswitch is initiated new calling to second callee, obtain second callee's media parameter, use re-invite message modification calling party's media parameter simultaneously, make the calling party and second callee set up conversation.
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| CN102185986A (en) * | 2011-04-06 | 2011-09-14 | 杭州华三通信技术有限公司 | Service registration method and equipment |
| CN103152495A (en) * | 2013-02-04 | 2013-06-12 | 华为终端有限公司 | Method, device and system for media transferring |
| CN103167053A (en) * | 2011-12-16 | 2013-06-19 | 中国移动通信集团公司 | Method and equipment for medium face address distribution in Internet protocol (IP) bearer establishment and soft switch system |
| WO2015131466A1 (en) * | 2014-03-04 | 2015-09-11 | 中兴通讯股份有限公司 | Data service processing method and device based on session initiation protocol (sip) |
| CN105491040A (en) * | 2015-12-07 | 2016-04-13 | 上海市共进通信技术有限公司 | Multiparty conference calling method based on SIP protocol |
| 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 |
| CN111756941A (en) * | 2020-06-23 | 2020-10-09 | 北京握联信息技术有限公司 | Method for controlling data interaction instruction by using INFO message based on SIP protocol |
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Cited By (10)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN102185986A (en) * | 2011-04-06 | 2011-09-14 | 杭州华三通信技术有限公司 | Service registration method and equipment |
| CN102185986B (en) * | 2011-04-06 | 2013-10-23 | 杭州华三通信技术有限公司 | Service registration method and equipment |
| CN103167053A (en) * | 2011-12-16 | 2013-06-19 | 中国移动通信集团公司 | Method and equipment for medium face address distribution in Internet protocol (IP) bearer establishment and soft switch system |
| CN103167053B (en) * | 2011-12-16 | 2016-06-29 | 中国移动通信集团公司 | IP carries the medium surface address distribution method in setting up, equipment and soft switchcall server |
| CN103152495A (en) * | 2013-02-04 | 2013-06-12 | 华为终端有限公司 | Method, device and system for media transferring |
| CN103152495B (en) * | 2013-02-04 | 2015-08-19 | 华为终端有限公司 | A kind of method of media transfer, Apparatus and system |
| WO2015131466A1 (en) * | 2014-03-04 | 2015-09-11 | 中兴通讯股份有限公司 | Data service processing method and device based on session initiation protocol (sip) |
| CN105491040A (en) * | 2015-12-07 | 2016-04-13 | 上海市共进通信技术有限公司 | Multiparty conference calling method based on SIP protocol |
| 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 |
| CN111756941A (en) * | 2020-06-23 | 2020-10-09 | 北京握联信息技术有限公司 | Method for controlling data interaction instruction by using INFO message based on SIP protocol |
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Application publication date: 20110330 |