KR20120110940A - Asdf - Google Patents

Abstract

PURPOSE: A communication method using a sip telephone for network address translation is provided to perform a call with a regular telephone connected to a public switched telephone network and different IP telephones in a public network using a single IP. CONSTITUTION: A private IP environment setting part is installed in an initial dialogue box. The private IP environment setting part comprises an enable button for setting a private environment, a use hello button for setting a dynamic IP environment in a private environment, and an IP address part. The enable button for a NAT(Network Address Translation) environment is checked in a static IP use state. An authorized IP allocated to a sharer is set. The authorized IP value of a preset IP sharer is converted into a 32-bit integer value of a global variable in a VOIP(Voice Over Internet Protocol) telephone program by pressing a save button.

Description

How to call SIP phone in private internet network {asdf}

The present invention relates to a call method for SIP in a private internet network, and a VIP environment using SIP.

Applicable IP router which uses IP router to solve IP shortage for the call from

Work to connect to different IP telephone and public switched telephone network (PSTN) in public network using one IP

The present invention relates to a call method for SIP in a private Internet network that enables a call with reversed communication.

The present invention relates to a call method for SIP in a private internet network, and a VIP environment using SIP.

Applicable IP router which uses IP router to solve IP shortage for the call from

Work to connect to different IP telephone and public switched telephone network (PSTN) in public network using one IP

The present invention relates to a call method for SIP in a private Internet network that enables a call with reversed communication.

In general, as the environment using the Internet is established, Internet phones are used. Use your existing Internet Phone (Voip)

The operating environment of FIG. 1 is a local area network (LAN) and a wide area network (WAN) as shown in FIG.

ork) and main area local area network (LAN). Local area network (LAN) is private IP environment and certified IP environment

The environment is separated through the IP router for the BIP (1). Private IP Environment IP Router

Japanese Patent Application Laid-Open No. 2003-0061242

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A connection is made between (1) and the VIP phone 2 through a private IP. Certified IP environment

The IP router (1) and the IP router (1) for the VIP to form a boundary

It converts and sends a set bipolar signal to take over to the router (3) and the Intenet Wide Area Network (WAN) that connect the call.

It consists of a digital service unit (4).

The main local area network (LAN) forms a certified IP environment, and the bipolar signal through the wide area network (WAN)

Digital service unit (4-1) for converting into unipolar signal and router (3-)

1) and the VIO exchanger 6 connected through the hub 5 to recognize the IP based on the passage through the router 3-1, and

A public telephone for connecting a call through the V IP gateway 7 and the V IP gateway 7 to the general telephone 8.

Switch PBX connected to the network PSTN, and the VIP phone 2-1 connected to the hub (5).

This VIP call method is to make a VIP call is made of two terminals (Voip Pho (Voip Pho phone)

ne)) (2,2-1) includes its IP address in the call signal and sends it to the opponent. IP address of signal source included in call signal

The signal receiver responds. In this case, if you are using a private IP, connect it to a VIO phone in a private IP environment.

One IP address is an unknown IP that is unknown to VIO phones in authorized IP environments.

The call answering party has a problem that call signal generation is impossible. In order to solve this problem, the IP router (1)

It was solved by replacing the existing equipment by making it separately. IP Router for VIO IP (1)

Regarding the call signal generated by the VIO IP phone 2, the IP router

The primary purpose of this is to convert and retransmit the IP address of the authorized IP address.

It can be recognized by IoIP phone.

However, if you are using an IP router for VIO IP, you need additional process for the unique function of IP router.

Delays and VoIP calls in the network are degraded. This is a common IP router is a private IP

It only performs the function of converting the address into a public IP address, but the IP router for VIO IP is connected via the IP router.

This is because it performs filtering on all IP packets and data updates on VIO signals.

In addition, it requires a layer change to the IP router structure, which shows the network layer in a common router environment.

2 is the same as FIG. This means that the VIO phone 2-2, the IP sharer 1-1, and the router 3-2 connecting to the Internet network are 5, 3, 3

It shows a layer structure.

FIG. 3 is a diagram illustrating a network layer in an IP router 1 environment for a VIO IP.

), The IP sharer 2, and the router 3 connecting to the Internet network form a 5, 5, 3 layer structure.

Therefore, the standard router of the general concept is that the IP sharer 1-1 proceeds in Layer 3 as shown in FIG.

IP sharer 1 for VIO IP is carried out in Layer 5 and requires additional overhead.

. This is a serious disadvantage from a network management perspective.

It also replaces the standard router with a router for VIO IP, but still incurs additional operating costs.

The compatibility with other standard routers can also cause problems. And you need to install a router for each VIP protocol.

Therefore, problems arise in maintenance and operation.

That is, the present invention provides an IP internet telephone, comprising: a private IP environment setting unit in an initial dialog box;

The private IP environment setting unit includes an enable button for private environment setting;

Japanese Patent Application Laid-Open No. 2003-0061242

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A youth hello button for setting a floating IP environment in a private environment;

It consists of an eye-dressing part which shows the eye-dresses which are designated by fixed / floating IP;

Fixed IP and floating IP can be selected and private IP environment setting part when using fixed IP (NAT Configuration)

Check the enable button (), set the authorized IP assigned to the router (NAT) (step a2), and press the save button.

If you set (step a3), the authorized IP value of the IP router is set to 32 in the global variable (natlp) in the VIO phone program.

Is converted to a bit integer value and stored, and the variable value useNat is set to (TRUE) (step a4) and then a VIP phone call.

Contact header of the time signal and the RTP connection address of the SDP body

IP phone for private Internet network, which uses IP as a variable (natlp) to communicate

It is to provide a way to find organic IP.

The present invention also relates to an SIP internet telephone, comprising: a private IP configuration unit in an initial dialog box;

The private IP environment setting unit includes an enable button for private environment setting;

A youth hello button for setting a floating IP environment in a private environment;

It consists of an eye-dressing part which shows the eye-dresses which are designated by fixed / floating IP;

When the call request signal (INVITE) is transmitted to the SIP proxy server which is the SIP exchange (S1), the SIP proxy server

Add a header (RECORD-ROUTE) containing the address of the scene to this signal and transmit it to the IP router (NAT) equipment (S2)

The router (NAT) refers to the address translation table it has and transmits it to the SIP phone of the private network (S3).

The telephone in the recipient's private IP network checks if it is in the private IP network.

Create a public IP header (CONTACT) owned by NAT router, and if it is a public IP network,

Calculate IP to generate header (CONTACT) and send a response signal to call setup request to SIP proxy server.

If the transmission (S4), and receives the SIP proxy server () is sent to the SIP phone in the public IP network (S5),

The sender's IP phone receives it and is added by the recipient's contact header and generated by intermediate nodes (proxy server).

Combined headers (RECORD-ROUTE) to generate the last header (ROUTE) in the order in which the header (RECORD-ROUTE) is generated

A final acknowledgment signal (ACK) is generated, and a final (FINAL) signal is transmitted along this path.

To provide a way to apply the IP of the IP router found to the IP signal

will be.

The present invention also provides a method of applying IP to SIP media information.

SIP used in the present invention is a protocol (SIP; Session Initation) used in the server (Voip Exchanger)

Protocol) is an Internet program defined and organized by the Internet Engineering Task Force (IETF).

A protocol for transmitting media (including telephones) over a protocol. It establishes, changes, and terminates sessions, such as Internet phone calls.

It is a protocol that can be used (Application Layer Protocol of OSI 7 Layer) and is being used as an existing Internet telephony service protocol.

Call setup time is shorter than standard (H.323) defined by ITU-T and various additional services are flexibly supported.

It is composed of technically superior architecture and consists of next generation internet telephone telephone service and IMT2000 service, and

It is a communication protocol that can be used as a protocol for additional services such as sex mailbox.

Hereinafter, embodiments of the present invention will be described in detail with reference to the drawings.

Figure 4 is a network configuration for carrying out the present invention, the usage environment of the VIP phone using the SIP server is local

Local area network (LAN), wide area network (WAN) and main area local area network (LAN).

Explain separately. Local area network (LAN) is divided into private IP environment and certified IP environment.

Each environment is separated through group 1-1. Private IP environment is a private network between IP Router (1-1) and SIP Phone (11).

It is configured to connect through IP. Certified IP environment includes IP Router (1-1) which is bounded by private IP environment,

Router (3) and Intenet Wide Area Network (WAN) for recognizing IP address through IP sharer (1-1)

It consists of a digital service unit (4) which is converted to a bipolar signal set to take over. The IP Router

(1-1) classifies data (TCP / IP) in Network Address Translation (NAT) to Layer 3 for transmission.

The IP router 1-1 shown in FIG. 2 can be illustrated.

The main local area network (LAN) forms a certified IP environment, and the bipolar signal through the wide area network (WAN)

Digital service unit (4-1) for converting into unipolar signal and router (3-)

1), the servers 6, 6-1, 6-2 connected through the hub 5 to recognize the IP based on the passage through the router 3-1, and the hub 5

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Switching to the SIP telephone 11-1, the general telephone 8 and the public telephone network PSTN connected through the

It consists of the SIP gateway 7 which is connected with.

In case they call VIO IP phone and VIO IP phone, they are located in the private IP network as shown in FIG.

If you make a call from a net phone (S-iPhone phone) to an Internet phone (S-iPhone phone) located in a certified IP network,

IP router (1-1) (IP for private Internet network (NAT)) to convert new IP into IP assigned to external authorized IP network

Call request is forwarded). IP router (1-1) is a client in private IP network managed by itself in the address table.

The private IP and port number for the client, the protocol number, and the recipient's IP and port number in the authorized IP network.

After registration, call request signal is sent to SPI proxy server 6-1 to resolve the receiver's exact IP address.

Send it. The STP proxy server 6-1 contains detailed location information of the SIP telephone managed by the STP.

Requests the exact location of the destination (recipient) IP phone from the location server and returns the information.

The call request signal is transmitted to the new SIP telephone 11-1. The receiving SPI phone receives the call request to accept the call request.

The call is made between the two phones by forwarding them in reverse order. At this time, please be careful

The destination sending the acceptance of the call is an IP in the private network (eg 192.168.1.101 in Figure 4), which is why

Since the location is unknown in the IP network, it is sent to the representative IP router (NAT). Share your accepted IP

NAT searches the IP information table of the private IPP client that it manages and places it in the private IPP.

Acceptance of a call is accomplished by securely forwarding an accept message for the call request to a correct SIP phone location.

This is followed by a flow of subsequent signals and message delivery between the sender and receiver as described above.

On the other hand, the call between the VIP phone and the public telephone network phone is performed as shown in FIG. This is different from that of FIG.

SIP Gateway (Media Gateway + Media Gateway Controller) at the location of conversion (SIP to SS7)

It is connected to the public telephone network through a protocol and media converter called SS7 Gateway. Specifically,

Once the private IP is located from the SIP telephone (11) located in the private IP network, the private IP is transferred to the external authorized IP.

The call request is transmitted to the IP router 1-1 which converts the IP assigned to the network. IP sharer (1-1) is connected to the address table.

Private IP and port number, protocol number, and authorized child for client in private IP network

After registering the IP and port number of the recipient in the bell pepper, it is used to resolve the receiver's exact IP address.

The call request signal is transmitted to the SIP proxy server 6-1. SPI proxy server 6-1 manages SPI

The accuracy of the destination (recipient) SIP telephone 11 to the location server 6 containing the detailed location information of the telephone 11.

A request for a location, the SIP location server 6 is a gateway (7) connected to the public telephone network to which the receiving telephone (8) is connected.

Information on the SIP proxy server 6-1 is transmitted. SIP proxy server 6-1 receives the call request signal

The inverter transmits a call request signal to the SPI gateway 7, which is a protocol converter with a public telephone network connected thereto. Eshua

IP gateway (7) receives this and converts SPI protocol to general telephone signal protocol (SS7)

When the call request signal is sent to the ventilation and the response to the call request is received from the exchange, the call request accept signal is received.

The call is made between the two phones by forwarding them in reverse order. At this time, please note that the receiving SIP telephone

The destination sending the acceptance of the agreement is IP on the private network (denoted as 192.168.1.101 in Figure 4).

Since the location is unknown in the IP network, it is sent to a representative IP router (NAT). Arc ball

Nat (1-1) searches the IP information table for the private IP network client that he manages

By stably forwarding the acceptance message for the call request to the correct SIP telephone 11 location located in the network,

Acceptance is made, and then the flow of subsequent signals is followed by a message between the sender and the receiver,

Will be delivered.

However, VIP phones have a problem that they use different information from the packets transmitted from the general Internet.

All. In other words, the general data only needs to transfer user data from the private IP to the destination through the Internet.

VIP uses the sender's information in the contents of the transmitted data for signal exchange.

This cannot be solved with the IP converter supported by the existing IP router.

The purpose of the transmission is to individually interpret signals such as VIO IP and to convert their contents accordingly.

It is outside the scope of performance of organic (NAT) and does not provide unless the product is developed and manufactured for a separate purpose. And

If it is manufactured separately for this, it is difficult to deal with various side effects such as packet delay. Then to support private IP

Find out what needs to be converted in VIO such as SIP.

First, the sender's address must be set to the public IP used by the IP router in the packet payload.

do.

Second, Real-Time Protocol Media Data Session Description Prot for sending and receiving media data

ocol) should be set to the public IP used by the IP router.

Third, if the IP used by the IP Router is a floating IP, it captures the Haiti change event on the terminal and

The actual authorized IP value of

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Fourth, the protocol supporting VIO IP recognizes that the sender of the received message is a device connected to a private network.

Therefore, it should be a protocol capable of transmitting the response to the received signal to the public IP assigned to the private network. In this case, the fourth is a table

It is outside the scope of the present invention by the support of the quasi-standardized SPI protocol.

7 is a view illustrating a solution of the private IP of the present invention, which appears when the setting button of the SIP phone is pressed.

Shown together with the screen, the private IP environment setting unit D2 is added.

Task 1 (How to find out the certified IP of the IP Router)

There are two ways to think about this, one is when the IP router is using a fixed IP, and the time or remote

IP distribution is flexible (eg ADSL) depending on the needs of the IP serving server.

Case 1 (when the IP Router is using a fixed IP),

In this case, a VIP phone and a private IP environment setting unit as shown in FIG. 7 and a profile diagram as shown in FIG. 8 will be described.

The enable check button of (NAT Configuration) which is the private IP setting part D2 of the right setting dialog box D1 of FIG.

Check (a1), set the official IP assigned to the router (NAT) (a2), and press the save button (D3) (a3)

The router's authorized IP value is converted to a 32-bit integer value in a global variable (natlp) in the VIO phone program.

(A4) and the variable value (useNat) is set to (TRUE) so that the contact header (Contact) of the call signal in the subsequent VIP phone call

Communication is performed using the variable (natlp) as the RTP connection address of the header and SDP body.

Is done.

Case 2 (if the IP router uses a floating IP),

In case that IP router that IP router uses fluidly like ADS is changed, terminal is called or periodically

The router generates a register signal to the proxy server to determine whether the IP value of the IP router changes.

Judging by comparing the received tag value in the Via header with the current IP router value

All. This case requires a slightly more complicated procedure than the previous case using a fixed IP.

As shown in Fig. 9, this is applied to the (NAT Configuration) which is the private IP setting unit 2 of the right setting dialog box D1 of Fig. 7.

Check the Enable Check Button and check the Use Hellow Button in (NAT Configuration) (Steps 1 and 2) (b1, b).

2) If you press the save button (step 3; b3), the official IP value of the IP router is

The global variable (usehellow) is saved with (TRUE) and the variable value (useNat) is set to (TRUE).

RTP Connecti of Contact Header and SDP Body of Call Signal on Phone Call

on) The communication is performed using a variable (usehellow) value as an address (step 4; b4).

SIP phones in the private network can check the IP information of the IP router or check the status of changes.

A signal (Register Query) is generated to the SIP proxy server, which is an IP exchange (step 5; b5). And the SPI response

Wait for coming (step 6; b6). The SPI proxy server sends a response to this request signal.

The Via header, which contains the certified IP of the IP router, the closest gateway that includes the parent path,

The received value of this header contains the latest authorized IP value of the IP router. Eshua Received This Signal

The IP phone calls a function (getIpFromVia (buffer, ip, amp; port)) to find the latest IP value of the IP router (step

7; b7) compare with a known value (step 8; b8). If the comparison is the same, no further procedure is required and the timer is reset

Set and wait for a predetermined time (step 10; b10). In other cases, the IP fluctuation has occurred, so the new child

By setting the public IP value of the IP router as the blood value, the SIP phone maintains the best value. after that

단말기 The IP terminal can be used to receive calls from the external telephone to the current telephone using the private IP.

After sending the signal for releasing the lock to SIP proxy server (step 11; b11), when a response signal is received, it is sent back to the latest information.

The registration signal is sent to the SIP proxy server (step 12; b12). And wait for a response (step 13; b13).

. Subsequently, when a response is received from the STP proxy server (step 14; b14), the IP value (NatIP) is reset to the modified IP (c.

In this dialog box, set the floating IP environment) (step 15; b15).

10 shows an example of a register signal of a request (left) and a response (right) for an SIP signal used in the present invention.

All.

Task 2 (Solution of how to recognize private IP address as authorized IP of IP sharer)

In VIO phones, there are two types of sessions between a sender and a receiver. One is to set up mutual calls

It is for signaling and the other is for media data transmission.

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The session for signaling for mutual call establishment depends mainly on signaling information of SIP protocol.

In a telephone using an IP, the sender represents the receiver representing the recipients in the private IP network for subsequent sessions.

Since the public IP address of the router (NAT) must be informed to the sender, this is a problem to be processed. Of this course

The concise expression may be represented as shown in FIG. 11. First, outgoing calls on the public network (the same is true for phones on private networks)

In order to set up a call to the Giga receiver, the call request signal (INVITE) is transmitted to the SPI proxy server 6-1, which is an SIP exchange.

When sending (S1), SIP proxy server 6-1 adds a header (RECORD-ROUTE) containing its address to this signal.

Transmit to IP router (NAT) equipment (S2). The IP Router (NAT) refers to its own address translation table.

Transmit to SIP telephone of private network (S3). The telephone in the private IP pepper that is the receiver is

If it is in a private IP network, the common IPRO header (CONTACT) that the IP router has.

If you create a public IP network, calculate your own local IP to generate a header (CONTACT)

The response signal for the call establishment request is transmitted to the proxy server (S4). The STP proxy server (6-1) receiving this information

When sending to the SIP phone in the IP network (S5), the sender SIP phone receives it and the contact of the receiver

By combining the header and the header (RECORD-ROUTE) generated and added by the intermediate nodes (proxy server), the final header (ROUTE)

) Are generated in the order in which the headers (RECORD-ROUTE) are generated, and the final acknowledgment signal (ACK) is generated.

Transmit signal (FINAL).

FIG. 12 is a flowchart illustrating message processing in a private api terminal, and is delivered via a router (NAT).

It is checked whether the received data is a request signal or a response signal (step 1; C1). In step 1; C1, the request type signal is

If it is a private IP network environment, it is checked if it is a private IP network (step 3; C3), or (step 5; C5).

A contact header is generated from the public IP value of the IP router NAT stored in the variable NatIP (step 3; C3). IP Router

(NAT) sends a response message to the SIP proxy server via the equipment (step 4; C4). Locale of VIO Phone

The address is calculated (step 5; C5), and a contact header is generated (step 6; C6).

Task 3 (solved how to enter SDP private caller address from private IP to authorized IP)

In a terminal using a private IP, a media transmission processes a message body in order to receive a transmission from a partner. on

In SIP, Invite, Ack, and OK signals on the call acceptor side at the call requester side

The characteristics of the media and network information (IP to be received) used by the user for the ATP session are shown as SDP (Session De).

in the field CONNECTION FIELD, SESSION IDENTIFIER FIELD,

The transmission allows point-to-point transmission from phone to phone. Must manage to support private IP

Is shown in FIG. 13.

In Fig. 13, when a signal is generated by the user in the VIO phone, it is determined whether the signal is a request or a response.

Check (Step 1). If the request is made in step 1, it is checked whether the request type is Invite (initial call request signal).

2). If it is byte in step 2, it is checked whether there is media information in the byte signal (step 3). Within the signal in (step 3)

If there is media information, set body set to True to change media information when sending ACK (final signal) later.

It is used to determine whether the exchange (step 4). Next, check whether the network environment of the VIO IP phone is private IP network.

(Step 5). If the private IP network in step 5, the connection information of the body is authorized to the IP router granted to the NAT

A message body including media information is generated (step 6), and a request signal is transmitted through the network (step 11).

). If it is not the invite (request) signal in step (2), it is checked whether it is an ack (final confirmation signal) (step 7). Axine in (step 7)

If it is an arc, check if the value of the bodyset is true (step 8); if it is true, send an Ack message (step 11);

Go to step 5). In step 5, if it is not a private IP network, it calculates its own authorized IP (step 9) to connect the media information.

The public IP is set in the information to generate a message body (step 10), and then a request message is transmitted (step 11).

FIG. 14 is an SDP processing flowchart of a receiving side, in contrast to FIG. 13, an SIP telephone in a private IP network is requested.

In the case of receiving a request signal, the S-IP VIO IP phone of the private IP network

Upon receiving the signal, it is checked whether the received signal request or response signal is performed (step 1). If the request is a signal, the request

Check whether the format is an byte (initial request signal) (step 2).

If the invite request is made in step 2, the receiving VIO phone is checked in the private IP network (step 3).

In the environment, the media information is generated by an authorized IP of the IP sharer (NAT) (step 4), and the receiver final response signal is transmitted (

Step 5), otherwise, calculate the IP of the SIP VIO IP phone (step 6), and calculate the media information with the calculated IP.

After generating (step 7), the receiver final response signal is transmitted.

The present invention described above is not limited to the above-described embodiments and drawings, and does not depart from the spirit of the present invention.

Various substitutions, modifications, and changes within the scope of the present invention are possible to those skilled in the art.

It will be obvious.

An object of the present invention is to set up the existing equipment of the VIP environment using SIP in the environment where the private Internet network is installed.

It is intended to provide a way to enable VIP calls without changing the policy.

To this end, the present invention provides a method of finding an IP of an IP router by an SIP phone for a private Internet network, and the found IP.

How to apply it to SPI signal on the basis of the information and how to apply it to SIP media information based on the found signal.

It is to give.

An object of the present invention is to set up the existing equipment of the VIP environment using SIP in the environment where the private Internet network is installed.

It is intended to provide a way to enable VIP calls without changing the policy.

To this end, the present invention provides a method of finding an IP of an IP router by an SIP phone for a private Internet network, and the found IP.

How to apply it to SPI signal on the basis of the information and how to apply it to SIP media information based on the found signal.

It is to give.

As described above, the present invention is a separate IP for IP without changing the existing network environment using a private IP network.

Eliminate the cost of replacing and introducing equipment by using an Internet phone without using a router.

Eliminates the overhead of the IP router compared to the Internet phone system using the existing VIP router.

As a result, the effect of eliminating quality degradation due to network load and voice data transmission delay can be expected.

And, in the case of VIP router, it was necessary to replace the router for each VIP protocol, but due to the present invention

It is expected to solve the compatibility problem of the IP sharing period by using the standard router as it is.

1 is a configuration diagram showing a call connection network of a typical VIO IP telephone,
2 is a diagram illustrating a network layer configuration in a general IP router environment;
3 is a diagram illustrating a network layer configuration in an IP router environment for VIO IP;
4 is a configuration diagram showing a usage environment of the present invention;
5 is a call diagram of the SIP telephone and SIP telephone of FIG. 4;
6 is a block diagram of a call between the SIP telephone and the general telephone of FIG. 4;
7 is a configuration example showing a dialog box for IP registration of the present invention;
8 is a flowchart of the IP phone of the present invention to find the IP of the IP router,
9 is a diagram showing a step of periodically recognizing the IP value change of the IP sharer used in the flow IP of the present invention
Row Chart,
10 is a view showing a relationship between a request signal and a response signal of a register signal for implementing the present invention;
11 shows a step of applying the IP of the IP router found by the SIP telephone of the present invention to the SPI signal.
Signal flow out,
12 is a flowchart illustrating a process of FIG. 11;
13 is applied to the IP side of the IP router found by the IP telephone of the present invention to the transmitting side of the IP media information.
Key is a flowchart representing a step,
14 is applied when the IP of the IP router found by the IP telephone of the present invention to the receiving side of the IP media information;
Key is a flowchart representing a step,
15 is an exemplary configuration diagram of an SIP signal and media information used in the present invention.

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Claims (1)

An SIP internet telephone, comprising: a private IP setting unit D2 in an initial dialog box D1;
The private IP environment setting unit D2 includes an enable button D21 for private environment setting;
Youth Hello button (D22) for setting the floating IP environment in a private environment;
An eye peer dress unit D23 representing an eye peer dress designated in the fixed / floating IP;
When using fixed IP, check enable button D21 of (NAT Configuration) which is private IP configuration unit (D2) (step a1),
After setting the authorized IP assigned to the router (step a2) and pressing the save button D3 (step a3), the set IP router
The authorized IP value of is converted to a 32-bit integer value in a global variable (natlp) and stored in the VIO IP Phone program.
The variable value useNat is set to (TRUE) (step a4), so that the contact header of the call signal (Contact Hea) in the VIO IP phone call.
der and RTP Connection address of SDP Body using variable (natlp) as IP
How to find the IP of the IP router, the IP router for the private Internet network characterized in that the communication.
[Claim 2]
An SIP internet telephone, comprising: a private IP setting unit D2 in an initial dialog box D1;
The private IP environment setting unit D2 includes an enable button D21 for private environment setting;
Youth Hello button (D22) for setting the floating IP environment in a private environment;
An eye peer dress unit D23 representing an eye peer dress designated in the fixed / floating IP;
When using floating IP, check the Enable check button and select Use Hellow Button of (NAT Configuration).
Check (steps 1 and 2) (b1 and b2) and press the save button (steps 3 and b3).
In the called program, the global variable (usehellow) is set to (TRUE) and stored (b4), and the variable value (useNat) is set to (TRUE).
In case of a VIO IP phone call, the contact header and the SDP body of the call signal are set.
Using a variable (usehellow) as an IP value as an RTP connection address;
SIP telephone for the private Internet network, characterized in that to perform the step (bb) to periodically recognize the change in the IP value
How to find the IP of the IP Router.
Claim 3.
3. The proxy server of claim 2, wherein the step (bb) of periodically recognizing a change in the IP value is an SIP exchanger.
Signal (Register Query), and waiting for the SPI response (step 6; b6),
The Roxy server sends back a response, which is the closest gate gateway that contains a rough path in the signal.
This includes the Via header, which contains the official IP of the IP router.
Since the IP router contains the latest authorized IP value of the router, the SPI phone that receives this signal has a function (getIpFromVia (buffer, i
p, amp; port)) to find the latest IP value of the IP router (step 7) and compare it with a known value (step 8),
If they are the same in comparison, no further procedure is required, the timer is reset and waits for some time (step 10),
In this case, since the IP change occurs, set the IP value of the IP router as the new IP value.
After that, the Epee IP terminal can receive a call from the outside to the current phone using the private IP.
Japanese Patent Application Laid-Open No. 2003-0061242
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A signal for releasing the existing registration information is sent to the SIP proxy server (step 11). When a response signal is received, the latest information is returned.
It sends a registration signal to SIP proxy server (step 12), and waits for a response (step 13).
When a response is received from the STP proxy server (step 14), the step of resetting the IP value (NatIP) to the modified IP (dial
SIP, which is an SIP exchanger, which is set to the floating IP environment in the log box) (step 15).
Generates a signal to the proxy server (Register Query) (step 5), waits for an SPI response (step 6), and
The proxy server sends back a response, which includes the nearest gate containing the rough path in the signal.
This includes the Via header, which contains the authorized IP of the Wayin IP Router. The received value of this header is
Since the IP router contains the latest authorized IP value, the SPI phone that receives this signal has a function (getIpFromVia (buffe).
r, ip, amp; port)) to find the latest IP value of the IP router (step 7) and compare it with a known value (step 8).
), If the comparison is the same, no additional procedure is required and the timer is reset and waited for some time (step 10), the other
In this case, since the IP fluctuation has occurred, set the common IP value of the IP router as a new IP value,
The best value is maintained by the SPI phone, and then the SPI terminal uses the private IP on the phone externally.
Sends a signal to the SIP proxy server to release the existing registration to receive the call to the current phone.
After stepping out (step 11), when a response signal is received, the registration signal is sent back to the SIP proxy server with the latest information (step 12), and
Waiting for a response (step 13), and then receiving a response from the STP proxy server (step 14), the IP value (Nat
IP) is reset to the modified IP (set to the floating IP environment in the dialog box) (step 15).
How to find the IP of the IP router, the IP phone for the private internet network.
[Claim 4]
An SIP internet telephone, comprising: a private IP setting unit D2 in an initial dialog box D1;
The private IP environment setting unit D2 includes an enable button D21 for private environment setting;
Youth Hello button (D22) for setting the floating IP environment in a private environment;
An eye peer dress unit D23 representing an eye peer dress designated in the fixed / floating IP;
When the call request signal (INVITE) is transmitted to the SIP proxy server 6-1, which is an SIP exchange (S1), the SIP proxy server
(6-1) adds a header (RECORD-ROUTE) containing its address to this signal and sends it to the IP Router (NAT).
(S2), the IP router (NAT) refers to the IP address table that it owns to the SIP phone of the private network.
Transmitting (S3), the phone in the recipient private IP network checks whether he is in the private IP network, and then in the private IP network
If it is in, create a public IP header (CONTACT) owned by the IP router (NAT), and in case of a public IP network
Calculate your local IP to generate a header (CONTACT), and use it to send a call setup request
Sending a response signal (S4), and receives the SIP proxy server (6-1) receives the SIP phone in the public IP network
(S5), the sender's SPI phone receives it and contacts the receiver's contact header and intermediate nodes (proxy server).
The headers (RECORD-ROUTE) generated and added are combined to generate the final headers (ROUTE).
It generates the final ACK signal ACK in order of creation, and transmits the final signal along this path.
This is a room to apply the IP of the IP router found by the IP router for the private internet network to the SIP signal.
method.
[Claim 5]
An SIP internet telephone, comprising: a private IP setting unit D2 in an initial dialog box D1;
The private IP environment setting unit D2 includes an enable button D21 for private environment setting;
Youth Hello button (D22) for setting the floating IP environment in a private environment;
An eye peer dress unit D23 representing an eye peer dress designated in the fixed / floating IP;
When a signal is generated by a user in a VIO phone, it is checked whether the signal is a request or a response.
If the request is made in steps 1; e1) and (step 1), it is checked whether the request type is Invite (initial call request signal) (step 2; e2).
, If it is byte in (step 2; e2), it is checked whether there is media information in the byte signal (step 3; e3), and signal in (step 3; e3).
If there is media information in the media set, the body set is set to True, and the media information is sent when the ACK is sent.
(E4), the network environment of the VIO IP phone is a private IP.
If it is a private IP network (step 5; e5) and (step 5; e5), the connection information of the body is assigned to the IP sharer (NAT).
As a public IP, a message body including media information is generated (step 6; e6), and a request signal is transmitted through a network.
(Step 11; e11), and if it is not an invite (request) signal in step (e2), check whether it is an acknowledge (final confirmation signal).
(7; e7), (7: e7), if the body signal is true, the value of the body set is true (step 8; e8).
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Message (step 11; e11), or go to (step 5; e5), if not private IPP in step 5; e5
After calculating the IP (step 9; e9), the public IP is set in the connection information of the media information to generate the message body (but
10; e10), the IP found by the SIP telephone for the private internet network performing a request message (step 11; e11).
How to apply the router's IP to the SIP signal.
Claim 6.
The method of claim 5, wherein
When a private IP network's SPI VIO IP phone receives a signal through the network, it receives a signal request.
Check whether the request type is in bytes (initial request signal).
(Step 2; f2), and if a request is received at (step 2; f2), it checks whether the receiving VIP phone is in a private IP network.
High (step 3; f3), if the private IP environment, the media information is generated by the authorized IP of the IP sharer (NAT) (step 4; f4), and the receiver
Transmit the final response signal (step 5; f5), otherwise calculate the IP of the SIP VIO IP phone (step 6; f6);
), After generating the media information with the calculated IP (step 7; f7), the private final response signal is transmitted.
How to apply IP router of IP router found by IP phone for internet network.
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