JP2006081202A - Message conversion device and IP telephone device - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide the message conversion apparatus and an IP telephone device in which the originator can make an outgoing call while concealing one's identity to a recipient, a telecommunications carrier having a SIP server can manage the originator by a conversion table, and an IP address of an IP telephone is not specified by a malicious third party by changing the IP address of the IP telephone at each outgoing call. <P>SOLUTION: The SIP server converts a SIP message of the originator and manages the conversion table. The IP telephone changes the IP address at each telephone call made by the IP phone. <P>COPYRIGHT: (C)2006,JPO&NCIPI

Description

  The present invention relates to a calling mode method when a caller calls a recipient in one-to-one communication on the Internet.

IETF RFC3261

IETF RFC2327 IETF RFC1889 Currently, with the rapid development of IP networks, there is an active movement to integrate switch-based telephone communication networks into IP networks. Each telephone carrier plans to transfer voice data over its own IP network using VoIP (Voice over IP). VoIP consists of two protocols: call signaling and session control protocol and voice transfer protocol. The Internet Engineering Task Force (IETF) has specified SIP (Session Initiation Protocol) (IETF RFC3261) for call signaling and session control. In addition, for example, SDP (Session Description Protocol) (IETF RFC2327) is applied to session descriptions such as voice compression schemes and transfer rate arrangements within SIP. The voice transfer protocol is not particularly determined by SIP, but RTP (Real-time Transport Protocol, described in RFC1889) is often used.

  According to IETF RFC3261, SIP sends and receives SIP messages consisting of the SIP start line, SIP message header, and SIP message body between the callers via the SIP server, calling the receiver, and the voice, video protocol, and bits during the call. This is a protocol for negotiating rates and the like between callers. The SIP start line mainly describes the action of the message sender, the SIP message header mainly describes the destination of the message receiver, the SIP server through which the message was passed, Call-ID (caller management number), etc., and the SIP message body Describes proposals for audio, video protocol, bit rate, etc. to be used during a call.

  In the following, the procedure from the start to the end of a two-party call by SIP described in IETF RFC3261 will be briefly shown, and the included issues will be described.

  FIG. 1 is a network diagram showing a two-party call using SIP.

  FIG. 2 is a sequence diagram showing the operation of a two-party call using SIP.

  In FIG. 1, UserA belonging to domain 3-1 and having IP phone 2-1 makes a call to UserB belonging to domain 3-3 and having IP phone 2-2 via SIP servers 1-1 to 1-3. .

  First, UserA sends a SIP message addressed to the start line INVITE and UserB to the SIP server A1-1 to make a call with UserB (11). The SIP server A1-1 that has received the INVITE message attaches the VIA header to the message header and transfers the SIP message to the SIP server B1-2. At that time, the SIP message of the start line 100 Trying is returned to the IP telephone set 2-1 as the SIP message transmission source (12). The SIP servers B1-2 and C1-3 that have received the SIP message perform the same operation and transfer the message to the IP phone 2-2 that User B has.

Upon receiving the SIP message, IP phone 2-2 sends a SIP message addressed to start line 180Ringing and UserA to SIP server C1-3 (13), and calls UserB with a bell or the like.
The SIP message of the start line 180Ring reaches the IP phone 2-1 via the SIP server.

  When UserB picks up the phone, IP phone 2-2 sends a SIP message addressed to start line 200OK and UserA (14), which reaches IP phone 2-1 via the SIP server. The IP phone 2-1 returns an ACK to the IP phone 2-1 (15), the voice packet passes through the main signal path 7, and the call starts (16).

  When the call is over, IP phone 2-1 of User A who ended the call sends a SIP message addressed to start line BYE and User B (17), reaches IP phone 2-2 via the SIP server, and ACKs for it The IP phone 2-2 sends (18), reaches the IP phone 2-1 via the SIP server, and the call is terminated.

  SIP is a protocol in which callers and receivers send and receive each other's SIP messages. The user name and domain name that can be received even if the receiver sends a reply to the sender, such as the From header indicating the sender in the message header, the To header indicating the recipient, and the Via header indicating the relayed SIP server (proxy mode) Describe etc. In addition, the caller describes the IP address or domain name of the own terminal in order to establish a session with the receiver by SIP.

  For VoIP, a method of calling that does not notify the receiver of the caller's identity (phone number, user name, etc.). First, the caller terminal creates a random user name, and then registers the random user name and the IP address of the terminal in the SIP server. A method of making the random user a source at the time of outgoing call is currently proposed in IETF RFC3261. In this method, the caller performs all steps of anonymous transmission.

  In the SIP system, if the caller proposed by the IETF hides his / her identity and communicates, only a random user name is registered in the SIP server, so the caller can send anonymous calls not only to the receiver but also to the SIP server. Will be going. Since the SIP server cannot determine the real user name from the user name registered in the server, it is difficult to control and manage outgoing calls. In an IP telephone service of a telecommunications carrier or the like, it is always necessary to manage users such as identification of callers, call availability management, and call duration management.

  In addition, if a malicious third party intercepts a SIP message, the SIP-URL and domain name of the sender and recipient that are described in the SIP message are found, which can cause damage such as nuisance calls. There is sex.

  Furthermore, if the IP address of the IP phone is known, a malicious third party may send a large number of packets to the IP phone after a call and may be subject to a DOS (Denial of Service) attack that interferes with device processing. Also gets higher.

  The purpose of the present invention is to perform "unknown transmission of transmission source" which hides the user name and affiliation domain of the SIP message sender from the receiver or malicious third party while the SIP server manages the caller And

  It is also intended to make it difficult for a third party to specify the IP address of an IP telephone and to minimize damage caused by knowing the IP address.

  One aspect of the present invention is a packet communication device for transmitting a message from a caller to a receiver, which has a function of converting and deleting at least a part of a message sent by the caller according to a request from the caller A processing unit having at least one and a control unit for determining whether or not to convert or delete at least a part of the message, and converting at least a part of the message in the processing unit based on the determination of the control unit Alternatively, it is a packet communication device to be erased. With such a configuration, it is possible to conceal information specifying the transmission source from the transmission destination.

As a specific example of at least a part of the message part to be converted or deleted,
(1) Of the SIP message header of the payload of the IP packet including the SIP message, a part for specifying the source user,
(2) A part for identifying a source domain in a SIP message header of an IP packet payload including a SIP message,
(3) Via tag part in the SIP message header of the payload of the IP packet including the SIP message,
(4) The part indicating the domain to which the Call-ID belongs in the SIP message header of the payload of the IP packet including the SIP message,
(5) The part representing the user name in the SIP message body of the payload of the IP packet including the SIP message,
One or a combination of these can be considered.

  The control unit may analyze the content of the message when receiving the message, and convert or delete the message part in the processing unit in response to the discovery of a specific character string or a specific header. The specific character string is a number that falls within the first few letters of the telephone number, for example, “184”. Further, the specific header is a SIP message header, and the message part may be converted or deleted in the processing unit triggered by the discovery of an extension header in the SIP message header.

  Furthermore, it is preferable to provide a table that stores correspondence before and after conversion of at least a part of the message. In order to conceal the sender's SIP-URL and domain, in an example of the present invention, a SIP server at a relay point from the sender to the receiver converts the SIP message in the SIP system. SIP server with message conversion function converts and deletes the outgoing message part of the caller, processing part that determines whether or not the message part is converted and deleted, and process that determines what information to convert based on And a table holding rules for conversion.

  In addition, in order to make it difficult for a third party to know the IP address of the IP phone that the caller has, the present invention generates and acquires a temporary IP address for the call as long as the IP message is sent, Dispose of the address. The IP phone generates and acquires an IP address in conjunction with SIP message transmission, discards the IP address when the call ends, a processing unit that registers the temporary IP address with the SIP server, and registers the SIP server with the temporary address when the call ends. It features a processing unit for canceling and a processing unit for generating a random interface ID.

  The SIP message conversion processing by the SIP server and the temporary IP address acquisition SIP message by the IP phone can be selected and used, or both or one of them can be used. Another aspect of the present invention is a communication method for an IP telephone, which includes a step of receiving a SIP message, a step of inspecting the SIP message to detect presence / absence of a non-notification transmission request, and a non- notification transmission request. In this case, it is a communication method for an IP telephone having a step of performing at least one process of change and deletion on at least a part of the SIP message and a step of transmitting the SIP message after the completion of the process. When there is a non-notification transmission request, it is desirable to perform a change process on at least a part of the SIP message and generate a table in which the correspondence before and after the change is recorded.

  Another aspect of the present invention is a communication method for an IP phone, in which a source address is changed from an original address to a temporary address at the start of a call, and the temporary communication is performed at the end of a call. This is an IP telephone communication method having a process of discarding a unique address.

  Another aspect is a method of converting or deleting a message part sent by a caller at the request of the caller, which includes determining whether to convert or delete the message part, It is characterized by a process for determining whether to convert based on such information and a rule for the conversion. In addition, it is a method to change the source for each outgoing call so that the user does not specify his / her outgoing source, and the process of changing the source in conjunction with outgoing calls and the use during the call at the same time as the call ends The telephone device is characterized by a process of disposing an address representing the identified identity, a method having addresses representing separate identities for sending and receiving, and a method of sending the user when performing the sending process. The scope of the present invention includes the above methods, apparatuses, and methods.

A function corresponding to the anonymous call function of the exchange telephone can be realized in the IP telephone.

<Example 1>
FIG. 3 shows a first embodiment of an IP telephone network using a SIP server with a message conversion function according to the present invention.

  FIG. 4 is a flowchart showing an operation procedure of the SIP server with message conversion function 12-1 in the present invention.

  FIG. 5 shows functional blocks of the SIP server with conversion function 12-1.

  FIG. 6 is a device configuration diagram of the SIP server 12-1 with a message conversion function.

  FIG. 7 shows an IP packet including a SIP message. 60 is an IP packet, 61 is an IPv4 / v6 header, 62 is a TCP / UDP header, 64 is a SIP start line, 65 is a SIP message header, and 66 is a SIP message body (SDP).

  The processing of the SIP server with conversion function 12-1 will be described below with reference to FIGS.

  First, the IP packet 60 shown in FIG. Next, the SIP message reception unit 45 performs SIP message reception 22, and the start line inspection 23, header inspection 24, and body inspection 25 are performed by the SIP message inspection unit 44. If there is an error in the SIP message, an error response notification 33 is made and the process ends.

  If there is no error, the message conversion presence / absence check 26 is then performed by the message conversion / processing unit 47.

  When there is no conversion request, the message conversion / processing unit 47 describes the Via header in the SIP message header 64 and transmits the SIP message from the IF 51. When there is a conversion request by a flag 561, 571 or the like (see FIG. 31) indicating that the sender is not notified, the message conversion / processing unit 47 performs the message header conversion process 27 and the message body conversion process 28. By this conversion process, the information indicating the transmission source becomes invisible from the transmission destination. After the conversion, conversion information extraction 29 is performed by the conversion information extraction / transfer unit 46 to extract header / body information necessary for generating the conversion table. Next, the conversion information table generation 30 is performed by the conversion entry generation / change unit 49, and the conversion information table write 31 is performed by the conversion entry input / output unit 48, and the conversion entry is registered in the conversion entry registration unit 50. The converted SIP message is transferred from the IF 51 by performing the message transfer 32 in the SIP message transmission / reception unit 45.

In the present embodiment, the functions of the conversion entry generation / processing unit 49 of the SIP message processing unit 41 and the conversion table processing unit 42 of FIG. 5 are executed by the CPU 72 of FIG. The functions of the conversion entry input / output unit 48 of FIG. 5 are executed by the conversion table calling unit 75 of FIG. The function of the conversion table registration unit 43 in FIG. 5 is executed by the conversion table storage unit 74 in FIG.
<Example 2>
Next, more detailed examples will be shown.

  FIG. 8 is a schematic diagram of a network configuration showing a second embodiment of the SIP server according to the present invention.

FIG. 9 is a sequence diagram showing a call procedure in a network example using the SIP server with a message conversion function in the present invention.
In the second embodiment, UserA makes a sender non-notification transmission to UserB. The non notification process is performed by the SIP server A12-2 with a conversion function. The SIP message conversion process is performed by the SIP server A (12-2) with a header conversion function to which the caller has transmitted a SIP message at the 111th and 112th steps in the sequence from the start to the end of the call shown in FIG. The operation of the SIP server with a conversion function is the same as in the first embodiment. 10, 11 and 12 show the conversion items of the SIP message in the second embodiment.

  FIG. 10 shows the header part (65 in FIG. 7) of the SIP message. The top shows before conversion and the bottom shows after return. In the second embodiment, the user name 142 of the From tag and the portion 143 representing the user of the SIP-URL in the header 141 before conversion are converted into character strings 147 and 148 that cannot specify an individual such as Anonymous, for example.

  FIG. 11 shows the body part (before conversion) of the SIP message. The body part of the SIP message is a part indicated by 66 in FIG.

  FIG. 12 shows the body part (after conversion) of the SIP message. Here, the part 152 representing the user name of the message body (SDP) 151 is converted into 156. Further, the portion 153 indicating the user address at the time of the call in the “o” and “c” tags is converted into an IP address when described in FQDN.

As described above, the information that the recipient can identify the sender, the user name 142 of the From tag, the part 143 that represents the user of the SIP-URL, the part 152 that represents the user name of the message body (SDP), etc. are converted, The sender information can be kept secret.
13 and 14 are conversion tables held by the SIP server 12 with a conversion function in the first embodiment. The conversion table converts all user names to the same one, and the real user name 171 and the Call-ID 172 associated table 170 or the anonymous user name 181 stored in advance in the SIP server, such as “Anonymous1-100”, are real. There is a table 180 or the like associated with the user name 182. Since the anonymous user name 181 and the Call ID 172 are associated with each other by this table, routing is performed without any trouble by referring to the table.
<Example 3>
FIG. 15 is a schematic diagram of a network configuration showing a third embodiment of the SIP server with message conversion function according to the present invention. Unlike the embodiment of FIG. 8, the SIP server B performs an anonymous process.

  FIG. 16 is a sequence diagram showing a call processing procedure in a network using the SIP server with a message conversion function in the present invention.

  In the third method, UserA performs transmission without notification to UserB. The SIP message conversion process is performed by the SIP server A (12-3) with a header conversion function that transmits a SIP message to the recipient at step 211 and step 212 in the sequence from the start to the end of the call shown in FIG. The operation of the SIP server with a conversion function is the same as in the first embodiment.

  FIG. 17 shows conversion items of the SIP message with message conversion function in the third embodiment. The SIP message header (65 in FIG. 7) before and after conversion is shown.

  In the third embodiment, the Via tag 222 in the header 221 before conversion is deleted, and only the Via tag 232 of the own server is described. For example, the user name 223 of the From tag is converted into a character string 233 that cannot identify an individual such as Anonymous, and the portion 224 representing the SIP-URL is converted into 225 in which the user name and domain name are not known. In addition, 225 indicating the domain to which the Call-ID belongs is converted to 235. The message body (SDP) conversion rules are the same as in the first embodiment.

  As described above, in the embodiment of FIG. 17, the estimation of the transmission source can be avoided by deleting the Via tag indicating the transit point of the message. In addition, by converting the part 225 indicating the domain to which the Call-ID belongs to a temporary domain name 235 in which the domain cannot be specified, estimation of the transmission source can be avoided.

FIGS. 18 and 19 are conversion tables held by the SIP server with conversion function 12-3 in the second embodiment. The conversion table converts all user names to the same one, and a table 271 associating the real user name 272, Call-ID 273, and domain name 274, or a user name stored in advance in the SIP server such as “Anonymous1-100”, for example. There is a table 281 that associates the user name 282, the real user name 283, and the domain name 284. Although routing is performed with reference to these tables, such information cannot be referred to at the reception destination, and the information of the transmission source is concealed.
<Embodiment 4> FIG. 20 is a schematic diagram of a network configuration showing a fourth embodiment of the SIP server according to the present invention.

  FIG. 21 is a sequence diagram showing a call processing procedure in a network using the SIP server with a message conversion function in the present invention.

  In the fourth method, UserA performs transmission without notification to UserB. The SIP message conversion processing is performed by the SIP server A with header conversion function (12-4) to which the caller has transmitted the SIP message at the 311 step and the 314 step in the sequence from the start to the end of the call shown in FIG. This is performed by the SIP server C (12-5) with a header conversion function that transmits a SIP message to the recipient at step 313. The operation of the SIP server with a conversion function is the same as in the first embodiment.

FIG. 22 shows conversion items of the SIP message in the fourth embodiment.
In the fourth embodiment, first, the SIP server 12-4 converts the user names 322 and 323 of the From tag in the header 321 before conversion into character strings 326 and 327 such as Anonymous, which are not identifiable, and the message body ( SDP) is converted according to the same rules as in the first embodiment. Next, in the SIP server 12-5, the Via tag 332 in the header 331 before conversion is deleted, and only the Via tag 336 of the own server is described. Further, the part 333 representing the SIP-URL of the From tag is converted to 337 whose domain name is unknown. In addition, the part 334 indicating the domain to which the Call-ID belongs is converted into 338.

  FIG. 23 shows a conversion table held by the SIP server with conversion function 12-4.

  FIG. 24 shows a conversion table held by the SIP server with conversion function 12-5.

The conversion table held by the SIP server 12-4 converts all user names into the same one, and stores the real user name 342 and the Call-ID 343 in association with each other in advance, such as the table 341 such as “Anonymous 1 to 100”. There is a table 351 that associates the saved user name 352 with the actual user name 353. The conversion table held by the SIP server 12-5 includes a table 361 in which the Call-ID 362 and the domain name 364 are associated with each other.
<Example 5>
FIG. 25 is a schematic diagram of a network configuration showing a fifth embodiment of the SIP server according to the present invention.

  FIG. 26 is a sequence diagram showing a call processing procedure in a network using the SIP server with a message conversion function in the present invention.

  In the fifth method, UserA makes a sender non-notification transmission to UserB. The SIP message conversion process is performed by the SIP server A with header conversion function (12-6) in the top level domain of each communication carrier at 411 step and 412 step in the sequence from the start to the end of the call shown in FIG.

FIG. 27 shows SIP message conversion items in the fifth embodiment.
In the fifth embodiment, the Via tag 421 is deleted from the header 420 before conversion, and only the description of 426 representing the Via tag of the own server, that is, the communication carrier is used. Also, the user name 422 of the From tag and the portion 423 representing the user of the SIP-URL are converted into character strings 427 and 428 that cannot identify an individual such as Anonymous, for example. In addition, 424 indicating the sender affiliation domain in the Call-ID tag is converted into the top level domain 429 of the communication carrier. The message body (SDP) conversion rules are the same as in the first embodiment. Also, the conversion table held by the SIP server with conversion function 12-6 in the fourth embodiment is shown in FIGS. 18 and 19, and the items of the table are the same as those in the second embodiment.
<Example 6>
FIG. 28 shows an operation flowchart of an IP telephone that uses a temporary IP address during a call.

  FIG. 29 is a functional block diagram of an IP telephone. The operation of the IP telephone according to the sixth embodiment of the present invention will be described below.

  FIG. 30 is an IPv6 address system diagram.

  FIG. 31 is a diagram showing a SIP message header when the sender is not notified.

  First, the operation at the start of a call will be described. When the caller decides to make a call to the other user, the IP telephone device 521 starts the SIP message sending process and performs the address acquisition method selection 502. When the random address generation is selected, the Rouer Solicitation transmission 503 is performed to acquire the IPv6 address prefix 551 from the router in the same subnet. When the router transmits a Router Advertisement for Router Solicitation, it performs Router Advertisement reception 505 and acquires an address prefix 551. Next, in order to determine the IPv6 interface ID 552, the interface ID generation 506 is performed by the random interface ID generation unit 528 to generate the interface ID 552. Examples of address prefixes and interface IDs are shown at 553 and 554 in FIG. Next, the IP address generation processing 507 is performed by the temporary IP address processing unit 529 using the address prefix 551 and the interface ID 552. When acquisition from the DHCP server is selected in the address acquisition method selection 502, an address acquisition request 504 is made to an external address distribution server such as a DHCP server to perform temporary IP address acquisition 508. When making an IP phone call using IPv4, be sure to obtain an address from an external server. Next, using the temporary address and the user name, the user data processing unit 532 generates a user information change entry or a new registration entry, and performs an account registration process 509. Next, the SIP message header generation unit 531 performs SIP message header processing 510, and the SIP message body generation unit 530 performs SIP message body generation processing 511.

  As shown in FIG. 31, when it is desired to make a call to the other party while hiding the caller, for example, the SIP message header 560 is prefixed with 184 in front of the other party's telephone number when making an unsolicited call in the exchange-based telephone system. , Such as a number (561) and an extension header (571), which indicate that the sender is not notified to the SIP server. Thereafter, the SIP signaling generation unit 525 generates a SIP start line INVITE indicating a request to the receiver, generates a SIP message, the IP packet processing unit 526 generates the IP packet 60, performs the SIP signaling transmission processing 512, and the call starts. To do. When the call ends, the user data processing unit 532 performs account registration deletion processing 514 to delete the account from the SIP server, and the temporary IP address processing unit 529 performs IP address discard processing 515 to complete the processing.

  In the case of SIP message reception, the process is the same as when starting, but temporary address acquisition and account registration are performed when the IP phone power is turned on, or when the IP phone enters the domain for which the SIP server is authorized, Receives a SIP message, makes a call, discards the temporary IP address at the end of the call, obtains a new temporary IP address immediately, and registers an account.

  As a result, the IP phone has two temporary IP addresses for sending and receiving SIP messages. However, in the case of IPv4, since two addresses cannot be set for one terminal, either for sending or receiving. With a temporary IP address of

By the SIP server with a header conversion function as an embodiment of the present invention described above, the SIP signaling receiver confirms the user name representing anonymity inside the converted SIP message, and the caller makes an anonymous call I can recognize that. If the domain name in the SIP message is hidden after being converted, deleted, the SIP message recipient cannot know the domain name of the sender. Also, when a malicious third party receives the caller's SIP message, the user name is hidden, making it difficult to identify the caller.
In addition, since the SIP server holds a conversion table related to the real user name and other parameters, it is possible to manage a caller such as a communication carrier.

  Even if a malicious third party eavesdrops on a SIP message and an IP packet during the call, the IP phone changes the IP address for each call by the IP phone using the temporary IP address of the embodiment of the present application. It becomes difficult to specify. Furthermore, when sending SIP messages using IPv6, there are a large number of IP addresses in the same segment, making it difficult to identify the sender by a third party.

It is a conceptual diagram showing the example of a SIP network created in order to demonstrate the subject of this invention. FIG. 6 is a sequence diagram showing an example of a call process in a SIP network created to explain the problem of the present invention. It is a conceptual diagram showing the SIP network in this invention. It is a flowchart figure which shows the operation | movement procedure of the SIP server with a message conversion function in this invention. It is a functional block diagram of a SIP server with a message conversion function in the present invention. It is an apparatus block diagram of a SIP server with a message conversion function in the present invention. It is a format figure which shows the IP packet containing a SIP message. It is a conceptual diagram which shows the example of a network using the SIP server with a message conversion function in this invention. It is a sequence diagram which shows the telephone call procedure in the example of a network using the SIP server with a message conversion function in this invention. It is a table | surface figure which shows the SIP message header after conversion by the SIP server with a message conversion function in this invention before conversion. It is a table | surface figure which shows the SIP message body (SDP) before the conversion by the SIP server with a message conversion function in this invention. It is a figure which shows the SIP message body (SDP) after conversion by the SIP server with a message conversion function in this invention. It is a table | surface figure which shows the conversion table which the SIP server with a message conversion function in this invention hold | maintains. It is a table | surface figure which shows the conversion table which the SIP server with a message conversion function in this invention hold | maintains. It is a conceptual diagram using the SIP server with a message conversion function in the present invention. It is a sequence diagram which shows the telephone call processing procedure in the network using the SIP server with a message conversion function in this invention. It is a table | surface figure which shows the SIP message header after conversion by the SIP server with a message conversion function in this invention before conversion. It is a table | surface figure which shows the conversion table which the SIP server with a message conversion function in this invention hold | maintains. It is a table | surface figure which shows the conversion table which the SIP server with a message conversion function in this invention hold | maintains. It is a conceptual diagram using the SIP server with a message conversion function in the present invention. It is a sequence diagram which shows the telephone call processing procedure in the network using the SIP server with a message conversion function in this invention. It is a table | surface figure which shows the SIP message header after conversion by the SIP server with a message conversion function in this invention before conversion. It is a table | surface figure which shows the conversion table which the SIP server with a message conversion function in this invention hold | maintains. It is a table | surface figure which shows the conversion table which the SIP server with a message conversion function in this invention hold | maintains. It is a network diagram using the SIP server with a message conversion function in the present invention. It is a sequence diagram which shows the telephone call processing procedure in the network using the SIP server with a message conversion function in this invention. It is a table | surface figure which shows the SIP message header after conversion by the SIP server with a message conversion function in this invention before conversion. It is a flowchart figure which shows from temporary address acquisition by the IP telephone in this invention to disposal. It is a functional block diagram of an IP telephone in the present invention. It is an IPv6 address system diagram. It is a table | surface figure which shows a SIP message header at the time of performing sender | caller unreported transmission.

Explanation of symbols

DESCRIPTION OF SYMBOLS 1 ... SIP server, 2 ... IP telephone, 3 ... Network domain, 11 ... IP telephone with a temporary address generation function, 12 ... SIP server with a converter function, 60 ... IP packet, 61 ... IPv4 / v6 header, 62 ... TCP / UDP header, 64 ... SIP start line, 65 ... SIP message header, 66 ... SIP message body (SDP), 72 ... General-purpose CPU, 73 ... General-purpose memory, 141 ... SIP message header, 15 ... 1 SDP, 170 ... Conversion table, 521 ... IP phone with temporary address generation function, 551 ... IPv6 address prefix, 552 ... IPv6 interface ID.

Claims (15)

A packet communication device for transmitting a message from a sender to a receiver,
A processing unit having at least one of a function of converting and deleting at least a part of a message transmitted by the caller according to a request of the caller;
A control unit that determines whether or not at least a part of the message is converted or deleted, and a packet communication device that converts or deletes at least a part of the message in the processing unit based on the determination of the control unit . At least a part of the message part to be converted or deleted is
(1) Of the SIP message header of the payload of the IP packet including the SIP message, a part for specifying the source user,
(2) A part for identifying a source domain in a SIP message header of an IP packet payload including a SIP message,
(3) Via tag part in the SIP message header of the payload of the IP packet including the SIP message,
(4) The part indicating the domain to which the Call-ID belongs in the SIP message header of the payload of the IP packet including the SIP message,
(5) The part representing the user name in the SIP message body of the payload of the IP packet including the SIP message,
The packet communication device according to claim 1, which is at least one of the following. The control unit
2. The packet communication apparatus according to claim 1, wherein when the message is received, the content of the message is analyzed, and at the time of discovery of a specific character string or a specific header, the processing unit converts or deletes at least part of the message part.   4. The packet communication apparatus according to claim 3, wherein the specific character string is a number included in the first few letters of a telephone number.   The certain character string is a number included in the first few letters of the telephone number, and the user name calculated from the telephone number is excluded from the first few letters and holds other telephone numbers and user names. 5. The packet communication apparatus according to claim 4, wherein the first few characters are removed when an inquiry is made to the apparatus and a message is transmitted.   4. The packet communication apparatus according to claim 3, wherein the specific header is a SIP message header, and the processing unit converts or deletes at least a part of the message part when an extension header in the SIP message header is found.   The packet communication device according to claim 1, further comprising a table that stores correspondence before and after conversion of at least a part of the message. An IP telephone communication method,
Receiving a SIP message;
Inspecting the SIP message to detect the presence or absence of an unsolicited call request;
When there is an unsolicited call request, performing at least one process of change and deletion on at least a part of the SIP message;
Sending the SIP message after the above processing;
A communication method for IP telephones. If there is an unsolicited call request, performing a change process on at least a part of the SIP message;
Generating a table that records the correspondence before and after the change;
The IP telephone communication method according to claim 8, comprising: An IP telephone communication method,
At the beginning of the call, changing the source address from the original address to a temporary address,
An IP telephone communication method including a process of discarding the temporary address at the end of a call. An IP telephone communication method,
Determining whether to perform random address generation;
When performing random address generation, obtaining an IPv6 address prefix from a router in the same subnet;
Generating an interface ID; and
Generating a temporary IP address from the IPv6 address prefix and the interface ID;
Using the temporary IP address and user name to generate a user information change entry or a new registration entry and perform an account registration process;
Steps to delete account registration and discard IP address when the call ends,
The IP telephone communication method according to claim 10, comprising:   11. The IP telephone communication method according to claim 10, wherein when performing an IP telephone using IPv4, an address is acquired from an external server and used as a source address.   11. The IP telephone communication method according to claim 10, wherein the source address is an IPv6 address, and an address having a random value is generated as the temporary address.   11. The IP telephone communication method according to claim 10, wherein the source address is an IPv6 or IPv4 address, the address is received from the address distribution server in conjunction with the first message transmission, and the address is released upon completion. The source address is an IPv6 address, and two separate addresses are set for message transmission and reception. The reception is disposable at the end of one call, and is generated immediately. 11. The IP telephone communication method according to claim 10, wherein the IP telephone communication method is disposed at the end of generation and call termination.

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