CN105515655B - A kind of voice main and standby rearranging method based on Session Initiation Protocol - Google Patents
A kind of voice main and standby rearranging method based on Session Initiation Protocol Download PDFInfo
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- CN105515655B CN105515655B CN201610031459.7A CN201610031459A CN105515655B CN 105515655 B CN105515655 B CN 105515655B CN 201610031459 A CN201610031459 A CN 201610031459A CN 105515655 B CN105515655 B CN 105515655B
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- 230000000977 initiatory effect Effects 0.000 title abstract 3
- 230000001360 synchronised effect Effects 0.000 claims abstract description 16
- 230000004044 response Effects 0.000 claims description 10
- 238000004891 communication Methods 0.000 claims description 6
- 230000011664 signaling Effects 0.000 claims description 6
- 239000003999 initiator Substances 0.000 claims description 3
- 230000008859 change Effects 0.000 abstract description 5
- 230000005540 biological transmission Effects 0.000 abstract description 2
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- 230000003111 delayed effect Effects 0.000 description 1
- 238000005516 engineering process Methods 0.000 description 1
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04B—TRANSMISSION
- H04B1/00—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission
- H04B1/74—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission for increasing reliability, e.g. using redundant or spare channels or apparatus
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04B—TRANSMISSION
- H04B10/00—Transmission systems employing electromagnetic waves other than radio-waves, e.g. infrared, visible or ultraviolet light, or employing corpuscular radiation, e.g. quantum communication
- H04B10/07—Arrangements for monitoring or testing transmission systems; Arrangements for fault measurement of transmission systems
- H04B10/075—Arrangements for monitoring or testing transmission systems; Arrangements for fault measurement of transmission systems using an in-service signal
- H04B10/079—Arrangements for monitoring or testing transmission systems; Arrangements for fault measurement of transmission systems using an in-service signal using measurements of the data signal
- H04B10/0791—Fault location on the transmission path
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04B—TRANSMISSION
- H04B10/00—Transmission systems employing electromagnetic waves other than radio-waves, e.g. infrared, visible or ultraviolet light, or employing corpuscular radiation, e.g. quantum communication
- H04B10/07—Arrangements for monitoring or testing transmission systems; Arrangements for fault measurement of transmission systems
- H04B10/075—Arrangements for monitoring or testing transmission systems; Arrangements for fault measurement of transmission systems using an in-service signal
- H04B10/079—Arrangements for monitoring or testing transmission systems; Arrangements for fault measurement of transmission systems using an in-service signal using measurements of the data signal
- H04B10/0793—Network aspects, e.g. central monitoring of transmission parameters
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- H—ELECTRICITY
- H04—ELECTRIC COMMUNICATION TECHNIQUE
- H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
- H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
- H04L65/1066—Session management
- H04L65/1101—Session protocols
- H04L65/1104—Session initiation protocol [SIP]
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Abstract
The invention discloses a kind of voice main and standby rearranging method based on Session Initiation Protocol, including：Master control is for electric on disk, transmission voice synchronizing signal after initialization；Master control master receives voice synchronizing signal, scans all voice ports, and log-on message, call information, Session Initiation Protocol stack key message and RTP channel informations are synchronized to standby disk in order；When master detect the voice log-on message of port either call information change or port in talk with an event is received under call state all the way, renewal master control for phase on disk log-on message or call information；When master control master failure or when restarting, the signal switched by master control for disk and master performs voice masterslave switchover.The present invention is using speech events driving and the active and standby synchronization of state change triggering, ensure that the synchronous real-time of speech business is high, masterslave switchover speed is fast, has no effect on being normally carried out for user speech business, allows the user for participating in speech business to perceive the presence less than failure.
Description
Technical Field
The invention relates to the field of optical communication voice, in particular to a voice main/standby switching method based on an SIP protocol.
Background
In an OLT (optical network terminal) communication system, a master control is a center of the entire communication system, and the master control generally adopts a 1+1 redundancy backup manner to ensure that a service on the OLT is not interrupted when the master control fails or encounters a restart, that is, when the working master control fails or encounters a restart, another spare disk which is idle can immediately become a master disk to work.
At present, the main/standby switching technology mainly aims at main/standby switching of hardware or main/standby switching of configuration and management, which rarely involves main/standby switching of voice services, but with the market and cost reduction requirements, it is now necessary to implement voice services on the main control of the OLT, so that when the main control fails to perform main/standby switching, the running voice services will be interrupted and suffer user complaints, and therefore the voice services also need to perform 1+1 software backup on the main control, which ensures that when the main control performs main/standby switching due to failure, the voice services can also perform main/standby switching, so that users participating in the voice services on the OLT cannot perceive the main/standby switching at this time of the main control.
Disclosure of Invention
The technical problem to be solved by the present invention is that when the master control performs the master-slave switching due to failure, the voice service can also perform the master-slave switching, and the user participating in the voice service on the OLT cannot perceive the master-slave migration at this time of the master control.
In order to solve the above technical problem, the technical solution adopted by the present invention is to provide a voice active/standby switching method based on an SIP protocol, which includes the following steps:
a10, electrifying a main control standby disk of the OLT communication system, and sending a voice synchronization signal after initialization;
step A20, after receiving the voice synchronization signal, the main control master scans all voice ports on the OLT, and synchronizes the registration information of each port, the SIP protocol stack key information of the port participating in the voice service, the call information of all dialogues and the RTP channel information to the standby disk in sequence;
step A30, when the main disk detects that the voice registration information or call information of the port changes, or one path of dialogue in the port receives an event in the call state, updating the corresponding registration information or call information on the main control backup disk;
step a40, when the running master control master disk fails or is restarted, the voice module receives the signal of switching the master control backup disk and the master control master disk, and executes the voice master/backup switching.
In the above-mentioned method, the first step of the method,
the registration information comprises registration refreshing of a registration timer or retry residual time after registration failure, a registration serial number, Call-Id and a registration state;
the key information of the SIP protocol stack comprises: the method comprises the steps that an INVITE signaling sent or received by a port which is stored in a main control main disc and participates in voice service relates to a key field of a conversation, and 180ring sent or received relates to the key field of the conversation;
the call information comprises call states of all conversations of a port participating in the voice service, a number graph and long-short timer matching state, a dialed number, a pick-up and hang-up state and a playback timer;
the RTP channel information comprises a far-end IP and a far-end RTP port number, and a local IP and a local RTP port number.
In the above method, the key field includes:
a from field, which is a logical flag of the request initiator;
a from-tag field, which is the local identification of the request originator;
the to field, which is the first and also the "logical" recipient of the first specified request;
a to-tag field, which refers to the local identification of the responder;
a call-id field, which is a unique mark for distinguishing a group of messages;
a via field identifying the address to which the response is sent back;
a branch-id field for distinguishing transactions requested to be created;
the contact field, which contains the contact information of the local terminal, usually consists of the user name and the full name of a host.
In the method, when the remaining time of the register timer and the playback timer synchronized to the main control standby disc are not 0, the register timer and the playback timer of the main control standby disc are started, but when the register timer or the playback timer of the main control standby disc is overtime, the specific operation is not executed.
In the above method, when the master subsequently sends or receives a 200OK termination response, the values of the key fields contact and to-tag related to the dialog in the 200OK are replaced with the values of the key fields contact and to-tag of 180 ring.
The invention adopts the voice event drive and the main-standby synchronization triggered by the state change, ensures the high real-time of the voice service software synchronization and the high speed of the main-standby switching, when the main disk has a problem or a fault, the voice service can immediately carry out the main-standby switching and can be quickly switched to the standby disk for processing, the normal operation of the voice service of a user is not influenced in the switching process, the user participating in the voice service can not sense the existence of the fault, and the method is also suitable for the main-standby switching of other voice protocol service software.
Drawings
Fig. 1 is a flowchart of a voice active/standby switching method based on an SIP protocol according to the present invention;
fig. 2 is a flow chart of real-time synchronization of voice services between a main control host and a backup disk according to an embodiment of a port as a calling party in the present invention.
Detailed Description
The invention is described in detail below with reference to the figures and specific examples.
The invention provides a voice active/standby switching method based on an SIP protocol, as shown in figure 1, comprising the following steps:
step A10, powering on a main control standby disk of the OLT communication system, and sending a voice synchronization signal to a running main control disk after initialization;
step A20, after the main control master disk receives the voice synchronization signal, it scans all voice ports on the OLT, and synchronizes the register information of each port, the call information of all CallLeg (dialog) of the port participating in the voice service, the SIP protocol stack key information of the port participating in the voice service and the RTP channel information to the standby disk in order, so that the voice key information of the standby disk and the master disk are kept consistent. Wherein,
the registration information comprises registration refreshing of a registration timer or retry residual time after registration failure, a registration serial number, a Call-Id (calling address), a registration state and the like, and the registration information forms a data structure and is synchronized to the main control standby disk; and if the remaining time of the timer synchronized to the main control standby disk is not 0 at this time, starting the registration timer, and when the registration timer of the main control standby disk is overtime, not executing specific operation, wherein the specific operation is executed by the running main control disk after the registration timer of the main control standby disk is overtime.
The call information includes the call state, number graph and long-short timer matching state of all CallLeg (dialogue) of the port participating in the voice service; the number dialed; the state of the on-off hook; playback timers, etc., for example, if one port has a three-party service, then there are two CallLeg (dialogs), then the call state, digit and length timer matching state, dialed number, pick-up state, playback timer, etc. of the two dialogs are synchronized to the master spare disk; if the playback timer synchronized to the main control spare disc is not 0 at this time, the playback timer is started, for example, if a port plays back a ring tone for 30 seconds and the total length is 60 seconds, 30 seconds remain, after synchronization, a playback timer for 30 seconds is started on the spare disc, and the specific operation is not executed after timeout, which is the same as the registration timer.
The key information of the SIP protocol stack comprises key fields of CallLeg (conversation) related to INVITE signaling sent or received by ports which participate in voice services and are stored in a buffer area of the conversation in a character string mode by a main control master disk, and key fields of conversation related to the conversation related to 180ring sent or received by the main control master disk;
key fields include from, from-tag, to-tag, call-id, via, branch-id, contact, etc.; the from field refers to the logical flag of the request initiator; the from-tag field refers to the local identification of the request originator; the to field refers to the first and also the "logical" recipient of the first named request; the to-tag field refers to the local identification of the responder; the call-id field is a unique mark for distinguishing a group of messages; the via field identifies the address to which the response is sent back; the branch-id field is used for distinguishing the transaction which is requested to be created; the contact field contains the contact information of the local terminal, and generally consists of a user name and the full name of a host. The main control backup disk for the key fields of the key information of the SIP protocol stack is obtained from the database and then the SIP protocol stack is regenerated, the SIP protocol stack creates CallLeg (dialog) and Transaction (things) which are the same as those of the main control backup disk according to the key fields, and therefore after the main and standby voice switching is executed, the signaling sent out subsequently by the port still belongs to the same session or the same things.
In the invention, since 180ring is a temporary response, is unreliable transmission, may be lost, and the key field contact is optional in 180ring, and may not carry the field in the INVITE signaling, when a 200OK termination response is subsequently sent or received, the value related to the session key fields contact and to-tag in 200OK is replaced with the value of the key fields contact and to-tag of 180 ring.
The RTP channel information comprises a far-end IP, a far-end RTP port number, a local IP, a local RTP port number and the like, and the RTP channel information is synchronized to the main control standby disk, so that the follow-up value-added service (such as three parties) can be ensured after the main control standby disk is switched, and the current local RTP port can not be reused by other ports.
Step a30, when the main disk detects that the voice registration information or call information of the port changes and/or a call leg (session) in the port receives a certain event in a certain call state, updating the corresponding registration information or call information on the main control backup disk, that is, synchronizing the relevant information of registration or call to the main control backup disk through the database channel, so that the state of each port of the main control backup disk is consistent with that of the main control main disk, but the main control backup disk does not really operate the voice service. The invention adopts two synchronization modes of state synchronization and event synchronization, the real-time state synchronization mode is the same as the batch synchronization mode of the power-on initialization of the standby disk, the data of the main control main disk is synchronized to the main control standby disk in real time, a CallLeg object and a Transaction object which are the same as the main control main disk are created according to the synchronous conversation key field, and a registration or playback timer is started.
Step a40, when the running main control main disk fails or is restarted, the voice module receives the signal of the main control backup disk and the main control main disk to switch, and executes the voice main/backup switching, and the main control backup disk executes the same voice service operation as the main control main disk, so that the voice service in progress is not interrupted and works normally.
The following describes a process of real-time synchronization of voice services between a main control main disk and a standby disk by taking a port as a calling party as an embodiment, and as shown in fig. 2, the specific process of real-time synchronization of voice services between the main control main disk and the standby disk in this embodiment is as follows:
when the user picks up the phone, the user hears the dial tone, at this time, the state is changed from the idle state to the dial tone state, the dial tone state and the duration of the dial tone timer are synchronized to the main control standby disk, and the standby disk starts the dial tone timer;
the user dials, the state changes from the state of the dial tone to the state of number detection at this moment, synchronize the state and dialing incident to the master control spare disk one by one, the master control spare disk cancels the dial tone timer;
when the number dialed by the main control main disk is matched with a certain figure, the number is reported through an INVITE signaling, at the moment, the state is changed into a calling state from a number detection change state, the state, calling information and SIP protocol stack key information are synchronized to the main control standby disk, and the main control standby disk creates a CallLeg (dialog) object and an INVITE (INVITE transaction in dialog) object which are the same as the main disk according to a dialog key field;
when the port of the main disk receives 180ring and then listens to the ring back tone, the state is changed from the calling state to the ring back tone state, the state and the time length of the ring back tone timer are synchronized to the main control standby disk, and the main control standby disk starts the ring back tone timer;
when the called user picks up the phone, the state is changed from ring back tone state to conversation state, the state and the RTP channel information related to the conversation are synchronized to the main control standby disk, the main control standby disk terminates the INVITE Transaction object, but keeps the CallLeg object;
after the synchronization, the state and data content of the master spare disk are basically consistent with those of the master main disk. When the running main control disk fails, the main control backup disk without failure replaces the main control disk to continue working, and at the moment, some business operations are performed according to the voice state and resources synchronized by the current main control backup disk, for example, when the port is in a state of registration failure, the port is re-registered after the timer is overtime, and the port is refreshed after the timer is overtime, such as the state of success. Therefore, when the main/standby voice switching is carried out, the interruption is avoided when the user listens to the dial tone, and the user continues to be in the state of listening to the dial tone; when the user is in a call, the user cannot perceive the abnormal change of the master control, and the call cannot be interrupted momentarily or delayed and the like.
It will be apparent to those skilled in the art that various changes and modifications may be made in the present invention without departing from the spirit and scope of the invention. Thus, if such modifications and variations of the present invention fall within the scope of the claims of the present invention and their equivalents, the present invention is also intended to include such modifications and variations.
Claims (4)
1. A voice main/standby switching method based on SIP protocol is characterized by comprising the following steps:
a10, electrifying a main control standby disk of the OLT communication system, and sending a voice synchronization signal after initialization;
step A20, after receiving the voice synchronization signal, the main control master scans all voice ports on the OLT, and synchronizes the registration information of each port, the SIP protocol stack key information of the port participating in the voice service, the call information of all dialogues and the RTP channel information to the standby disk in sequence;
step A30, when the main disk detects that the voice registration information or call information of the port changes, or one path of dialogue in the port receives an event in the call state, updating the corresponding registration information or call information on the main control backup disk;
step A40, when the main control main disk running fails or is restarted, the voice module receives the signal of the main control standby disk and the main control main disk to switch, and executes the voice main/standby switch;
the registration information comprises registration refreshing of a registration timer or retry residual time after registration failure, a registration serial number, Call-Id and a registration state;
the key information of the SIP protocol stack comprises: the INVITE signaling sent or received by the port participating in the voice service and stored in the main control main disk relates to a key field of a conversation, and the sent or received temporary response 180ring relates to a key field of a conversation;
the call information comprises call states of all conversations of a port participating in the voice service, a number graph and long-short timer matching state, a dialed number, a pick-up and hang-up state and a playback timer;
the RTP channel information comprises a far-end IP and a far-end RTP port number, and a local IP and a local RTP port number.
2. The method of claim 1, wherein the key field comprises:
a from field, which is a logical flag of the request initiator;
a from-tag field, which is the local identification of the request originator;
the to field, which is the first and also the "logical" recipient of the first specified request;
a to-tag field, which refers to the local identification of the responder;
a call-id field, which is a unique mark for distinguishing a group of messages;
a via field identifying the address to which the response is sent back;
a branch-id field for distinguishing transactions requested to be created;
the contact field, which contains the contact information of the local terminal, usually consists of the user name and the full name of a host.
3. The method of claim 1, wherein the main standby disc registration timer and the playback timer are started when a remaining time of the registration timer synchronized to the main standby disc and the playback timer are not 0, but a specific operation is not performed when the registration timer or the playback timer of the main standby disc is expired.
4. The method of claim 2, wherein when the master subsequently transmits or receives the termination response 200OK, the values of the key fields contact and to-tag related to the dialog in the termination response 200OK are replaced with the values of the key fields contact and to-tag of the temporary response 180 ring.
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Citations (7)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN101110862A (en) * | 2006-07-18 | 2008-01-23 | 中兴通讯股份有限公司 | Speech processing system implementing method |
| CN101640611A (en) * | 2009-08-26 | 2010-02-03 | 烽火通信科技股份有限公司 | Method for synchronous and real-time reversion for VOIP voice service primary and standby modules |
| CN102111459A (en) * | 2009-12-28 | 2011-06-29 | 中兴通讯股份有限公司 | Call maintenance method and device in active/standby switching of IP voice equipment |
| CN102185753A (en) * | 2011-01-30 | 2011-09-14 | 广东佳和通信技术有限公司 | Device for realizing dual-backup switching of Ethernet link inside communication equipment |
| CN102404189A (en) * | 2010-09-15 | 2012-04-04 | 杭州华三通信技术有限公司 | Main and standby switching method of device supporting multi-topology routing as well as master control board |
| CN103944697A (en) * | 2014-03-27 | 2014-07-23 | 广东佳和通信技术有限公司 | Double-link double-server voice dispatching system and redundant backup method |
| CN104410526A (en) * | 2014-11-28 | 2015-03-11 | 华为技术有限公司 | Call control method, Diameter protocol forwarding equipment and system |
Family Cites Families (1)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN1921478B (en) * | 2005-08-26 | 2011-09-14 | 华为技术有限公司 | Business signal transmitting method based on network agreement |
-
2016
- 2016-01-18 CN CN201610031459.7A patent/CN105515655B/en active Active
Patent Citations (7)
| Publication number | Priority date | Publication date | Assignee | Title |
|---|---|---|---|---|
| CN101110862A (en) * | 2006-07-18 | 2008-01-23 | 中兴通讯股份有限公司 | Speech processing system implementing method |
| CN101640611A (en) * | 2009-08-26 | 2010-02-03 | 烽火通信科技股份有限公司 | Method for synchronous and real-time reversion for VOIP voice service primary and standby modules |
| CN102111459A (en) * | 2009-12-28 | 2011-06-29 | 中兴通讯股份有限公司 | Call maintenance method and device in active/standby switching of IP voice equipment |
| CN102404189A (en) * | 2010-09-15 | 2012-04-04 | 杭州华三通信技术有限公司 | Main and standby switching method of device supporting multi-topology routing as well as master control board |
| CN102185753A (en) * | 2011-01-30 | 2011-09-14 | 广东佳和通信技术有限公司 | Device for realizing dual-backup switching of Ethernet link inside communication equipment |
| CN103944697A (en) * | 2014-03-27 | 2014-07-23 | 广东佳和通信技术有限公司 | Double-link double-server voice dispatching system and redundant backup method |
| CN104410526A (en) * | 2014-11-28 | 2015-03-11 | 华为技术有限公司 | Call control method, Diameter protocol forwarding equipment and system |
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