CN102263775B - Method and device for controlling local session initiation protocol (SIP) calling - Google Patents
Method and device for controlling local session initiation protocol (SIP) calling Download PDFInfo
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- CN102263775B CN102263775B CN201010189071.2A CN201010189071A CN102263775B CN 102263775 B CN102263775 B CN 102263775B CN 201010189071 A CN201010189071 A CN 201010189071A CN 102263775 B CN102263775 B CN 102263775B
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Abstract
The invention provides a method and a device for controlling local session initiation protocol (SIP) calling. Network equipment in an area detects whether an SIP server is available; when the SIP server is unavailable, an INVITE message of a calling party is broadcast in the local network; and establishment of the SIP calling is controlled according to a received corresponding response message. The network equipment can forward the INVITE message to a corresponding virtual local area network (VLAN); the SIP network equipment can be an SIP gateway, an SIP user agent, an SIP terminal or the like. Thus, when the SIP server is unavailable, a certain function of the local SIP calling also can be maintained in a local network range.
Description
Technical field
The present invention relates to communication technical field, particularly relate to a kind of one's respective area SIP calling-control method and device thereof.
Background technology
Ip voice, as a kind of means of communication of high performance cheap, is just subject to the favor of increasing enterprise customer in worldwide.The existing conventional voip solution based on SIP (Session initiation Protocol) is all the framework based on client-server (Client-Server), and the speech ciphering equipment of each branched structure is all utilize the sip server being positioned at a central point to complete the functions such as Call-Control1, call routing, user management.
Usually, in order to ensure to communicate when a sip server (we are called " main sip server ") breaks down and can normally carry out, main sip server is taken over by arranging backup sip server, example as shown in Figure 1, main sip server and backup sip server are generally man-to-man, when main sip server normally works, each SIP gateway (or sip terminal, the speech ciphering equipments such as sip user agent) send message to main sip server, backup sip server can not receive the information of any terminal, when SIP gateway breaks down at main sip server, it sends registration message to backup sip server and back up sip server provides Call-Control1, call routing, the services such as user management.But, due to characteristics such as regions, once region 1, network connection interruption between 2 and central field, region 1,2 all ip voices will be in paralyzed state immediately.
Exemplified by legend 2, although operator and SIP voice service provider can arrange two other extra backup sip server 2,3 respectively in region 1,2, to maintain SIP calling continuity within the scope of one's respective area during network connection interruption in region 1, between 2 and central field, but this solution needs extra cost and server maintenance, cost performance is not high.
Summary of the invention
The present invention is intended to solve aforementioned one or more technical problem, provides a kind of SIP one's respective area Call-Control1 technical scheme, in the disabled situation of sip server, can maintain certain SIP one's respective area call function within the scope of localized network.
According to an aspect of the present invention, the method controlling one's respective area SIP and call out is provided in a kind of network equipment here, comprises the steps: a), detect sip server and whether can use; B), when sip server is unavailable, in the network model of one's respective area, broadcast INVITE (request) message of calling subscriber; C), according to received corresponding response message, control this SIP call establishment.
Abovementioned steps b) in, described INVITE can be forwarded to corresponding VLAN (VLAN) by the network equipment, and/or described INVITE is performed local loop back operation.
Preceding method comprises the INVITE that the network equipment receives broadcast in the network range of one's respective area further, checks whether requested object is present networks equipment institute service-user.
According to another aspect of the present invention, a kind of network equipment that can be used for one's respective area SIP Call-Control1 being provided here, comprising: checkout gear, whether can use for detecting sip server; Dispensing device, when sip server is unavailable, broadcasts the INVITE of calling subscriber in the network model of one's respective area; Receiving system, for receiving corresponding response message; Processing unit, controls this SIP call establishment according to described response message.
Described INVITE is forwarded to corresponding VLAN by aforementioned dispensing device, and/or described INVITE is performed local loop back operation.
Aforementioned receiving system can receive the INVITE of broadcast in the network range of one's respective area further, and control device checks whether requested object is present networks equipment institute service-user.
The present invention has following technical advantage:
In the disabled situation of primary, spare server, the SIP basic call business of low limit can be realized in certain area, to deal with urgent emergency situations.
Technology realizes simple, only needs simply to change basic SIP call flows does some in related network device.
With low cost, because its function i ntegration is on existing equipment, do not need for a standby server using probability minimum is separately established in each region, regular maintenance expense does not also at all increase.
Accompanying drawing explanation
By the detailed description with the accompanying drawing proposed below, feature of the present invention, character and advantage will become more obvious, and element identical in accompanying drawing has identical mark, wherein:
Fig. 1 is one of a kind of typical SIP network architecture example;
Fig. 2 is two of a kind of typical SIP network architecture example;
Fig. 3 is SIP network architecture according to an embodiment of the invention example;
Fig. 4 is SIP one's respective area according to an embodiment of the invention call flow schematic diagram.
Embodiment
Below in conjunction with accompanying drawing, the preferred embodiment of the present invention is described in detail.
Schematic network structure example of the present invention as schematically shown in Figure 3, it comprises three networks being positioned at zones of different scope, wherein area 0 is as central station, is provided with active and standby part sip server and controls management such as the Call-Control1 between the SIP gateway in each region, call routing, user's registration and certifications; Area 0,1,2 networks can be Layer2 switching networks, wherein SIP gateway 1,2,3,4,5 can be POTS (analog telephone business) terminal provides the analog phone line interface of standard, supports multiple popular encoding and decoding speech, and other network function etc.Each SIP gateway maintains the connection with active and standby part sip server by periodically sending the mode such as REGIST (registration) message or OPTION (option) message; After the dialing of calling subscriber's off-hook, the SIP gateway of its correspondence sends INVITE order to sip server, this INVITE order is after sip server analyzing and processing, be sent to the SIP gateway that called subscriber is corresponding, afterwards, caller SIP gateway obtains the response of called SIP gateway to set up the calling of asking between calling subscriber and called subscriber.
Here, we illustrate for SIP gateway 4,5, lose communication link between area 0 or standby usage sip server when losing efficacy in region 2, communicate between each user terminal maintaining inside, region 2 by realizing the present invention.
As previously mentioned, whether SIP gateway 4 can be used by some sip message is detected sip server as heartbeat, such as, periodically transmission REGIST message or OPTION message are to primary, spare sip server, do not receive reply in time and can think that sip server is out of touch unavailable.
When active and standby sip server is neither available, SIP gateway 4 enters broadcast mode, broadcasts the INVITE that SIP gateway 4 is initiated itself in the network range of one's respective area 2.Usually when network design, can by voice-and-data delineation of activities to the upper transmission of different VLAN (such as Voice VLAN and data vlan), so that arrange qos policy, based on this, we can be forwarded to corresponding Voice VLAN by aforementioned INVITE, so, all SIP network equipments in the network range of region 2 can receive this message.Particularly, after user 1 off-hook dial-up user 2, SIP gateway 4 broadcasts an INVITE by Voice VLAN, and this message comprises
r-URIthe details of (requested uniform resource identifier), voice flow agreement and the All Media coding etc. of support thereof, and start retransmission time out clock T1.
If SIP gateway 4 receives any 100Trying response message, this shows that one's respective area has the network equipment to receive INVITE and INVITE will be sent to destination according to routing iinformation by it, SIP gateway 4 stops retransmission time out clock T1, waits for 18x or 200 response messages;
If SIP gateway 4 receives the 180RING message that the SIP gateway 5 from called subscriber 2 correspondence is replied, it sends ringing-current sound to calling subscriber 1; Further, if SIP gateway 4 receives the 200OK response message that the SIP gateway 5 from called subscriber 2 correspondence is replied, it sends ACK acknowledge message to SIP gateway 5;
If SIP gateway 4 receives 4xx/5xx/6xx response message, reply ACK immediately to responder, but these INVITE affairs (Transaction) are not terminated on SIP gateway 4, continue to wait for 18x or 200 response messages, (do not support the present invention to prevent the sip user agent that has in the network range of region 2 or sip terminal until retransmission time out clock T1 time-out, and the INVITE of the non-user served for this sip user agent or sip terminal is made to the response of refusal, cause INVITE affairs to be terminated).
In order to ensure also to converse between the user that coexists on SIP gateway 4, local loop back operation can be carried out to INVITE, adopt SIP gateway 4 of the present invention can send a go back to this locality when INVITE being sent to network simultaneously, its object IP address is 127.0.0.1, to ensure also can converse between the user that coexists on SIP gateway 4.
Below, how we carry out call treatment further combined with SIP gateway 5 when receiving the INVITE of broadcasting in the network range of local zone 2: as previously mentioned, it receives the SIP INVITE of broadcast by Voice VLAN, check the user profile whether R-URI in this message meets this SIP gateway and serve: if the INVITE of the user served for this SIP gateway, then according to normal INVITE handling process, send ring to called subscriber 2 and reply 180RING response message to source-SIP gateway 4, 200OK response message is replied further when called subscriber 2 off-hook replies this calling, represent and successfully accepted and processed INVITE request, and by this message of response by the IP address of self, port numbers, payload type, the information such as the medium encoder that payload type is corresponding send SIP gateway 4 to, if not the INVITE of the user served for this SIP gateway, this message will be dropped, and not have other to operate.
And when SIP gateway 4,5 is when detecting that sip server can be used, it switches back normal mode, follow-up all INVITE will no longer can send in a broadcast mode, and the sip message of SIP gateway to all broadcast modes received of normal mode all directly abandons, and is left intact.
What deserves to be explained is, the SIP gateway 4,5 in legend 3 also can be can be sip user agent or independently sip terminal equipment, or the network equipment of integrated aforementioned functional device.
Fig. 4 is SIP local call schematic flow sheet according to an embodiment of the invention, and we are under aforementioned active and standby part sip server down state here, and the flow process performing SIP local call is described:
Step S401, user 1 terminal is a common POTS terminal, carries out dial-up as calling party's off-hook.
Step S402, SIP gateway 4 is when this POTS off-hook call being detected, and now it is broadcast mode, and in the network range of one's respective area 2, broadcast an INVITE, this message comprises R-URI, the details of voice flow agreement and the All Media coding etc. of support thereof; SIP gateway 4 can utilize SIP gateway supported vlans agreement, and aforementioned INVITE is forwarded to corresponding Voice VLAN.
Step S403, SIP gateway 5 is receiving the INVITE from Voice VLAN in the network range of local zone 2, check whether called subscriber is this equipment institute service-user, it is by checking whether the R-URI in this INVITE meets this SIP gateway institute service subscriber information, if for this SIP gateway institute service-user, it sends ring to called subscriber 2; On the contrary, if not for this SIP gateway institute service-user, this INVITE will be dropped.
Step S404, SIP gateway 5 judge called subscriber for this equipment institute service-user time, it replys 180RING response message further to SIP gateway 4, represents that it has received this message, points out called subscriber, such SIP gateway 4 would not time-out and abandon.
Step S405, SIP gateway 4 is receiving the 180 RING message of replying from SIP gateway 5, and it sends ringing-current sound to calling subscriber 1.
Step S406, SIP gateway 5 detects called subscriber 2, and whether off-hook is replied;
Step S407, SIP gateway 5 replys 200OK response message to SIP gateway 4 further when called subscriber 2 off-hook replies this calling, represent and successfully accepted and processed INVITE request, and send the information such as the IP address of self, port numbers, payload type, medium encoder that payload type is corresponding to SIP gateway 4 by responding this message.
Step S408, SIP gateway 4 replys ACK, confirms to have received to the final response of INVITE request and by the medium encoder all supported according to two gateways and carry out speech coding and Streaming Media transmission.
So far, the SIP connection setup between calling subscriber 1 and called subscriber 2.
Be illustrated as although above-mentioned and the invention provides some embodiments; not be used for limiting protection scope of the present invention; the professional of the art without departing from the scope and spirit in the present invention, can carry out various amendment to embodiment, and this amendment is all within the scope of the present invention.
Claims (7)
1. control the method that one's respective area SIP (Session initiation Protocol) calls out in the network equipment, it is characterized in that comprising the steps:
A). detect sip server and whether can use;
B). when sip server is unavailable, in the network range of one's respective area, broadcast INVITE (request) message of calling subscriber, described INVITE is forwarded to corresponding VLAN (VLAN) or described INVITE is performed local loop back operation by the network equipment;
C). according to received corresponding response message, control this SIP call establishment.
2. the method for claim 1, described step a) in access network device periodically send logon message and detect sip server by its response message and whether be in available.
3. the method for claim 1, it comprises steps d further): the INVITE receiving broadcast in the network range of one's respective area, checks whether requested object is present networks equipment institute service-user.
4. can be used for a network equipment for one's respective area SIP Call-Control1, it is characterized in that comprising:
Checkout gear: whether can use for detecting sip server;
Dispensing device: when sip server is unavailable, INVITE (request) message of calling subscriber is broadcasted in the network range of one's respective area, wherein, described INVITE is forwarded to corresponding VLAN (VLAN) by dispensing device, or described INVITE is performed local loop back operation;
Receiving system: for receiving corresponding response message;
Processing unit: control this SIP call establishment according to described response message.
5. whether the network equipment as described in right 4, is characterized in that described checkout gear periodically sends logon message and detects sip server by its response message and be in available.
6. the network equipment as described in right 4, is characterized in that described receiving system receives the INVITE of broadcast in the network range of one's respective area, and described control device checks whether requested object is present networks equipment institute service-user.
7. the network equipment as described in right 6, is characterized in that the described network equipment can be sip terminal, sip user agent or SIP gateway.
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| CN201010189071.2A CN102263775B (en) | 2010-05-31 | 2010-05-31 | Method and device for controlling local session initiation protocol (SIP) calling |
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| CN103825868B (en) * | 2012-11-19 | 2017-12-22 | 华为技术有限公司 | A kind of method, local gateway and the system of local voice escape |
| CN105790903A (en) * | 2014-12-23 | 2016-07-20 | 中兴通讯股份有限公司 | Terminal and terminal call soft handover method |
| CN110113623B (en) * | 2019-04-18 | 2021-07-27 | 浙江工业大学 | Audio and video slice transmission platform based on SIP protocol |
| CN110266712B (en) * | 2019-06-28 | 2022-02-01 | 苏州会信捷信息科技有限公司 | Communication method combining mobile internet emergency and common method |
| CN110971768B (en) * | 2019-12-06 | 2020-12-25 | 深圳震有科技股份有限公司 | SIP call processing method and system, computer equipment and medium |
| CN112073266B (en) * | 2020-09-03 | 2022-04-05 | 北京珞安科技有限责任公司 | Instruction distribution system and method based on heartbeat mechanism |
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| CN1949781A (en) * | 2005-10-10 | 2007-04-18 | 阿尔卡特公司 | System and method for establishing emergency communications in a telecommunication network |
| CN101106524A (en) * | 2006-07-14 | 2008-01-16 | 日立通讯技术株式会社 | Packet transfer device and communication system |
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| Publication number | Priority date | Publication date | Assignee | Title |
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| CN101552952B (en) * | 2009-05-06 | 2011-12-07 | 中兴通讯股份有限公司 | A broadcast call implementation method and system |
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| CN1949781A (en) * | 2005-10-10 | 2007-04-18 | 阿尔卡特公司 | System and method for establishing emergency communications in a telecommunication network |
| CN101106524A (en) * | 2006-07-14 | 2008-01-16 | 日立通讯技术株式会社 | Packet transfer device and communication system |
| CN101668097A (en) * | 2009-09-21 | 2010-03-10 | 杭州华三通信技术有限公司 | Escaping method for IP telephone set and communication device |
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Address after: 201206 Pudong Jinqiao Export Processing Zone, Nanjing Road, No. 388, Shanghai Patentee after: Shanghai NOKIA Baer Limited by Share Ltd Address before: 201206 Pudong Jinqiao Export Processing Zone, Nanjing Road, No. 388, Shanghai Patentee before: Shanghai Alcatel-Lucent Co., Ltd. |
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