TWI664847B - Sip gateway, call origination method thereof and call termination method thereof - Google Patents

Abstract

The invention provides a session initiation agreement gateway, a speaking method and a receiving method thereof. After the session initiation protocol gateway interrupts communication with the session initiation protocol server, if a call is initiated, the called number is encapsulated into a query broadcast packet to inquire about the session initiation protocol gateway to which the called user equipment belongs. The session initiation protocol gateway to which the called user equipment belongs returns its own address in response to this inquiry broadcast packet, so that the calling user equipment and the called user equipment can establish a call through a point-to-point session initiation protocol. With this, even if the connection is disconnected from the session initiation protocol server, the user equipment can still talk to each other.

Description

Session initiation agreement gateway, speaking method and receiving method thereof

The present invention relates to a Session Initiation Protocol (SIP), and in particular, to a session initiation protocol gateway, a calling method and a receiving method thereof.

At present, the development of Voice over IP (VoIP) networks is mainly based on SIP, and the communication system architecture of SIP mainly includes a SIP server and SIP terminal equipment. SIP terminal equipment can be SIP phones, SIP gateways and other equipment. The user communicates through the SIP terminal device, and the SIP terminal device is registered with the SIP server, and with the assistance of the SIP server, the end-to-end communication of the Wide Area Network (WAN) is performed.

When the communication between the SIP terminal device and the SIP server is interrupted, the SIP terminal device cannot communicate with other terminals on the network. In addition, if the terminal is a SIP gateway, the number of users under its jurisdiction can reach hundreds of users, and if the communication is interrupted, the communication between hundreds of users under its jurisdiction will be impossible. Considering that the same computer room may be equipped with multiple SIP gateways at the same time, these SIP gateways serve thousands and tens of thousands of users in a region, and these users may also make emergency calls such as 110/119. The interruption of communication with the SIP server will cause the communication of tens of thousands of users in the area to stop, and emergency telephones such as 110/119 will not be available, which will have a wide range of impacts.

To solve this problem, in the prior art, when a single gateway is disconnected from the SIP server, communication between users under the gateway itself can be maintained. However, users between the gateways still cannot communicate. Or, statically configure the data of all users in the local area on each SIP gateway to maintain local area communication services. However, static configuration users have a large workload and face slow data update, which cannot reflect the current configuration status of users in a timely manner. Furthermore, a local SIP server is set up in the local area network so that when the network is interrupted, the function of communication services between the local gateways can be maintained. However, this mechanism changed the network topology, and the additional local server required additional management and cost burden.

It can be seen that there are still many shortcomings in the above-mentioned customary methods. It is not a good design, and it needs to be improved.

In view of this, the present invention provides a SIP gateway, a method of speaking and a method of receiving the same, which are to improve the shortcomings of the aforementioned existing SIP gateway.

The speech method of the SIP gateway of the present invention includes the following steps. Determine whether the SIP gateway has lost communication with the SIP server to enter the local survival mode. In the local survival mode, the SIP gateway encapsulates the called number dialed by the calling user device into a query broadcast packet, and transmits the query broadcast packet in a broadcast manner, and the query broadcast packet is used to find the SIP to which the called user device belongs. Gateway. A call between the called user equipment and the calling user equipment is established in a Peer-to-Peer (P2P) SIP manner.

The receiving method of the SIP gateway of the present invention includes the following steps. Receive an inquiry broadcast packet, and the inquiry broadcast packet encapsulates the called number dialed by the calling user equipment. Determine whether the called number is about the user to which the SIP gateway belongs. If the called number is about the user to which the SIP gateway belongs, reply to the Internet Protocol (IP) address of the SIP gateway. Accept the call between the calling user equipment and the called user equipment corresponding to the called number established in the P2PSIP manner.

The SIP gateway of the present invention includes a sending unit, a receiving unit, a broadcasting unit, a security unit, a user unit, a number unit, and a processing unit. The sending unit sends the encapsulated IP packet to the IP network. The receiving unit receives an IP broadcast packet. The broadcasting unit encapsulates the number of the called user equipment in the inquiry broadcast packet, and sends the inquiry broadcast packet by broadcast in the IP area network to inquire the IP address of the SIP gateway to which the called user equipment belongs, and accepts the reply. The result of the inquiry. The security unit encapsulates the security token in the transmitted packet, and analyzes whether the received packet carries the security token. The user unit determines whether the called user equipment is a user under the jurisdiction of the SIP gateway. The number unit converts the called number into a format conforming to the SIP protocol. The processing unit executes a P2PSIP signal processing flow.

Based on the above, by querying the SIP gateway to which the called user equipment belongs by broadcast, it can learn the IP address of the SIP gateway when the communication with the SIP server is interrupted, and enable the calling user equipment and the called party. User equipment can use P2PSIP to talk.

In order to make the above features and advantages of the present invention more comprehensible, embodiments are hereinafter described in detail with reference to the accompanying drawings.

Please refer to FIG. 1, which is an application system architecture diagram according to an embodiment of the present invention. The system architecture includes a SIP gateway 1, a SIP server 4, and a user equipment 6 (for example, a phone and a mobile phone supporting the SIP protocol). As can be seen from the figure, multiple SIP gateways 1 are set up in the local area network 2. Generally, after the SIP gateway 1 is registered with the SIP server 4 through the wide area network 3, it can be provided by the SIP server 4. The user equipment 6 under the jurisdiction performs all calling behaviors (including calls between the user equipment 6 under the same SIP gateway 1).

When the line connected to the WAN 3 is abnormal (for example, the external IP access line 5 is abnormal), the communication of the SIP gateway 1 is almost suspended. For example, the external and local calls of the SIP gateway 1 cannot be established. Therefore, the local survival mechanism must be activated so that user communication between all SIP gateways 1 on the local area network 2 can be maintained.

2 is a block diagram of components of a SIP gateway 1 according to an embodiment of the present invention. The SIP gateway 1 includes a sending unit 11, a receiving unit 12, a broadcasting unit 13, a security unit 14, a user unit 15, a number unit 16 and Processing unit 17.

Sending unit 11: responsible for sending the encapsulated IP packet to the IP network.

The receiving unit 12 is responsible for receiving an IP broadcast inquiry packet.

Broadcasting unit 13: Responsible for encapsulating the called number of the called user equipment 6 in the IP query broadcast packet, and inquiring the IP address of the SIP gateway 1 of the called user equipment 6 by broadcasting in the IP local area network, and accepting Respond to the results of the inquiry.

Security unit 14 (optional): Responsible for establishing a mechanism for trusting secure packets, allowing a security-identified security token (Token) to be encapsulated in a packet sent in a local survival mode, and analyzing whether the received packet carries a recognizable security Token, so that the SIP gateway 1 in the group can identify whether the source of the packet can be trusted. This security token can use standard or custom mechanisms, such as ID identification, password, key, and so on.

User unit 15: In the local survival mode, before SIP gateway 1 sends a call request message or receives a call request message, it will first pass through user unit 15 to confirm whether the called user equipment 6 of this request is a SIP gateway. 1 Users under their jurisdiction.

Number unit 16: Responsible for converting the emergency service phone (for example, 110/119, etc.) dialed by the user into the corresponding local phone number, and converting the local phone number dialed by the calling user device 6 into a format conforming to the SIP agreement For example, the format used by E.164 SIP registered accounts containing country code and area code). If the number dialed is a non-local call such as long distance, mobile, etc., the call request from the calling user device 6 is directly rejected. On the receiving end, the received E.164 format number is converted into a local telephone number format that the user is familiar with, and the user's phone is displayed.

Processing unit 17: responsible for monitoring the status of the network between SIP gateway 1 and SIP server 4. When the condition of disconnection between SIP gateway 1 and SIP server 4 is established, the local survival mode of SIP gateway 1 is started; , Responsible for the Peer-to-Peer (P2P) SIP signal processing flow.

It should be noted that the aforementioned sending unit 11, receiving unit 12, broadcasting unit 13, security unit 14, user unit 15, number unit 16 may be software modules and may be executed by the processing unit 17 or may be specific chips. , Circuit, processor and other hardware components. The processing unit 17 may be a central processing unit (CPU), or other programmable general purpose or special purpose microprocessor (Microprocessor), digital signal processor (DSP), programmable controller, special application product Body Circuit (ASIC) or other similar components or a combination of the above. The SIP gateway 1 can be replaced with a SIP terminal in any form (for example, a SIP phone, a SIP video phone, a SIP soft phone, etc.).

FIG. 3 is a flowchart of a local survival call initiated by the SIP gateway 1 according to an embodiment of the present invention. The method includes:

Step 701: The SIP gateway 1 determines that the communication with the SIP server 4 is interrupted according to the criteria to enter the local survival mode. This criterion may be conditions such that the response signal from the SIP server 4 is not received, the packet from the SIP server 4 is not received for a period of time, the registration fails, or the IP packet drop rate exceeds a preset value, and so on.

Step 702: After the SIP gateway 1 receives the calling number of the calling user device 6, it checks whether it is a 110/119 emergency call or a local call; if it is an emergency call, it is directly converted to the corresponding local call; if it is Long-distance calls (not emergency calls or local calls) will directly reject the calling request of the calling user equipment 6. In this embodiment, the local survival method of the SIP gateway only deals with the scenario of local calls.

Step 703: The number unit 16 of the SIP gateway 1 converts the collected number into the E.164 format used for SIP registration. For example, the 4245557 number dialed by the user is converted into the E.164 format +88634245557 containing the country code and area code.

Step 704: Compare the recipient number in the E.164 format with the user's SIP account information on SIP gateway 1, and check whether the called user equipment is connected to the same SIP gateway 1 as the calling user equipment.

Step 705: If it is on the same SIP gateway 1, the SIP gateway 1 directly accepts the call request and rings the user equipment 6 directly. After the called user equipment 6 answers, the call can be established.

Step 706: Since the registration authentication mechanism has been lost at this time, if the calling and receiving user equipment 6 are not on the same SIP gateway 1, the security unit 14 encapsulates the security for security identification in the SIP message of the call request. Token provides SIP gateway 1 in local area network 2 to determine whether the received packet is a packet sent by SIP gateway 1 in a trusted group, that is, group identification.

Step 707: The SIP message encapsulated with the security token is further encapsulated into an IP packet in a broadcast format (that is, an inquiry broadcast packet), and the inquiry broadcast packet is transmitted in a broadcast manner to ask whether the called number is within the local area network 2. Users accommodated by SIP gateway 1. It should be noted that the broadcast format can be SIP, Address Resolution Protocol (ARP), or other standard protocols and custom protocols that can use the broadcast IP address to transmit the called number.

Step 708: If a response is received from other SIP gateways 1 in the local area network 2, the processing unit 17 starts a P2PSIP call establishment process instead of the mechanism provided by the SIP server 4.

It should be noted that after the initial call request packet uses the broadcast IP address as the destination IP address, after receiving the response, the IP address of the SIP gateway of the user equipment 6 to which the call was received is changed to The IP address of the SIP gateway of the user equipment 6 to be called is the destination IP address to perform the subsequent SIP call establishment process. On the other hand, if the non-SIP protocol is used to query the IP address of the SIP gateway to which the called user device 6 belongs, the IP address of the SIP gateway to which the called user device 6 belongs is obtained before the use of the called The IP address of the SIP gateway to which the device 6 belongs is the destination IP address and starts to send the SIP initial call request packet.

FIG. 4 is a flow chart of receiving a local survival inquiry broadcast packet received by the SIP gateway 1 according to an embodiment of the present invention. The method includes:

Step 801: Receive an inquiry broadcast packet inquiring the location of the called user equipment 6 in a broadcast manner.

Step 802: Check and query the security token in the broadcast packet to determine whether it is a trusted packet in the group.

Step 803: If the inquiry broadcast packet sent by the trusted SIP gateway 1 in the group, check whether the called number in the inquiry broadcast packet is a user under jurisdiction.

Step 804: if the user is under the jurisdiction, return the IP address of the SIP gateway 1 itself in response to the query broadcast packet.

Step 805: Start a P2PSIP call establishment process.

Step 806: If the untrusted packet and the called number are not under the jurisdiction of the subscriber number, then the packet is directly discarded.

To sum up, the embodiment of the present invention uses the packet broadcast technology. After the SIP gateway detects that the communication with the remote SIP server is interrupted, the broadcast mode is activated. It asks for a broadcast packet to find the location of the SIP gateway where the locally called user equipment is located. After receiving the inquiry broadcast packet, the SIP gateway to which the called user equipment belongs analyzes the called number as the user under the jurisdiction, and then responds to the inquiry IP address of the broadcast packet itself. Other SIP gateways will discard the packets after receiving the inquiry broadcast packet and analyzing the users whose gate numbers are not under their jurisdiction. After detecting the location of the SIP gateway where the locally called user equipment is located, the SIP gateway to which the calling user equipment belongs no longer sends the call message to the SIP server, but instead sends the calling request message directly to the called The SIP gateway (through P2PSIP) to which the device belongs, completes the subsequent call establishment procedures.

Features and effects

The local survival mode of the SIP gateway provided by the embodiment of the present invention has the following advantages when compared with the cited case and other conventional technologies:

1. In the embodiment of the present invention, it is not necessary to set up an additional SIP server on the local area network to provide communication when the SIP gateway is connected to an external network. It has the consistency of the network topology and maintains the network configuration settings. Advantages of change.

2. The embodiment of the present invention breaks through the limitation of a single SIP gateway, so that all SIP gateways on the same local area network can maintain call communication with each other when the external network is abnormal.

3. The embodiment of the present invention is provided with a checking mechanism for trusted packets to avoid interference from packets in non-group devices.

4. The embodiment of the present invention can be applied to a covered application of an exchange office to provide emergency related telephone services in the event of an abnormal external network in a region.

5. The embodiment of the present invention eliminates the tedious work of statically setting the user information of the neighboring SIP gateway, and automatically finds the SIP gateway IP address of the receiver by querying and learning.

6. Provide a security mechanism for trusted device authentication to prevent attacks from unknown sources in local survival mode.

7. There is no need to change the network architecture. Under the existing network architecture, a disconnection with the remote SIP server is detected, and the local survival mechanism is automatically activated.

Although the present invention has been disclosed as above with the examples, it is not intended to limit the present invention. Any person with ordinary knowledge in the technical field can make some modifications and retouching without departing from the spirit and scope of the present invention. The protection scope of the present invention shall be determined by the scope of the attached patent application.

1‧‧‧SIP Gateway

2‧‧‧ Local Area Network

3‧‧‧ WAN

4‧‧‧ WAN

5‧‧‧external IP access line

6‧‧‧user equipment

11‧‧‧ sending unit

12‧‧‧Receiving unit

13‧‧‧broadcasting unit

14‧‧‧Security Unit

15‧‧‧Customer Unit

16‧‧‧ number unit

17‧‧‧ processing unit

701 ~ 708, 801 ~ 806‧‧‧ steps

FIG. 1 is a structural diagram of an application system according to an embodiment of the present invention. FIG. 2 is a block diagram of components of a SIP gateway according to an embodiment of the present invention. 3 is a flowchart of a SIP gateway according to an embodiment of the present invention. FIG. 4 is a flowchart of a SIP gateway receiving call according to an embodiment of the present invention.

Claims (10)

A speaking method of a Session Initiation Protocol (SIP) gateway includes: determining whether a SIP gateway interrupts communication with a SIP server to enter a local survival mode; in the local survival mode, the The SIP gateway encapsulates the called number dialed by a calling user device into an inquiry broadcast packet, and transmits the inquiry broadcast packet in a broadcast manner, wherein the inquiry broadcast packet is used to find a SIP gateway to which the called user device belongs. And establishing a call between the called user equipment and the calling user equipment in a Peer-to-Peer (P2P) SIP manner according to the called number dialed by the calling user equipment. A method for receiving a SIP gateway includes: receiving an inquiry broadcast packet, wherein the inquiry broadcast packet encapsulates a called number dialed by a calling user device; judging whether the called number is about a user to which a SIP gateway belongs; if The called number refers to the user to which the SIP gateway belongs, replying to the Internet Protocol (IP) address of the SIP gateway; according to the IP address of the SIP gateway corresponding to the called number To accept a call between the calling user equipment and the called user equipment corresponding to the called number established in a P2PSIP manner. The calling method according to item 1 of the scope of patent application, wherein the local survival mode further includes: checking whether the called number is an emergency phone or a local phone number; if it is an emergency phone, directly calling the phone number Into a corresponding local call; and if it is not the emergency call or the local call, the call request from the calling user device is rejected. The calling method according to item 1 of the scope of patent application, wherein before transmitting the inquiry broadcast packet by broadcast, the method further comprises: determining whether the called user equipment and the calling user equipment are connected to the same SIP gateway; and If so, ring directly to the called user equipment. According to the calling method or the receiving method described in item 2 of the scope of the patent application, the inquiry broadcast packet is further encapsulated with a security token (Token) for group identification. The speech method according to item 1 of the scope of patent application, wherein after transmitting the inquiry broadcast packet in a broadcast manner, the method further comprises: after the initial speech request packet uses the broadcast IP address as the destination IP address; and After obtaining the IP address of the SIP gateway to which the called user equipment belongs, the IP address of the SIP gateway to which the called user equipment belongs is used as the destination IP address for the subsequent SIP call establishment process. The method of speaking according to item 1 of the scope of patent application, wherein after transmitting the inquiry broadcast packet in a broadcast manner, the method further includes: if using a non-SIP protocol to inquire the IP address of the SIP gateway to which the called user equipment belongs, to be obtained After the IP address of the SIP gateway to which the called user equipment belongs, the IP address of the SIP gateway to which the called user equipment belongs is used as the destination IP address to start sending SIP initial call request packets. The speaking method according to item 1 of the scope of patent application, wherein determining whether the SIP gateway interrupts communication with the SIP server includes determining whether a response from the SIP server has not been received within a preset period of time, or whether an IP packet is dropped. Whether the rate exceeds the preset value. A SIP gateway includes: a sending unit that sends the encapsulated IP packet to an IP network; a receiving unit that receives an IP broadcast packet; a broadcasting unit that encapsulates the number of a called user equipment into an inquiry broadcast In the packet, the inquiry broadcast packet is sent by broadcast in the IP local network to inquire the IP address of the SIP gateway to which the called user equipment belongs, and accept the result of replying to the inquiry; a security unit, a security Token is encapsulated in the transmitted packet and analyzes whether the received packet carries the security token; a user unit determines whether the called user equipment is a user under the jurisdiction of the SIP gateway; a number unit converts a called number into A format that complies with the SIP protocol; and a processing unit that executes a P2PSIP signal processing flow based on the called number that conforms to the SIP protocol format. The SIP gateway according to item 9 of the scope of patent application, wherein in the P2PSIP signal processing flow, the processing unit sets the destination IP address of the SIP packet to the SIP gateway learned by broadcast IP address.