JP2009239564A - Sip telephone exchange system and sip telephone exchange method - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To connect an all-purpose SIP instrument in an IP-PBX which cannot provide SDP information to "INVITE" as well, and to prevent a sound breakdown. <P>SOLUTION: An RTP repeating installation 104 for controlling and repeating Media for an RTP is arranged, so that when "INVITE" having no SDP information is received from an IP-PBX 103, the RTP repeating installation 104 decides Media information univocally (M1, M2) to rewrite the SDP information to M2, and "INVITE" having the SDP information is sent to a VoIP terminal 102. Further, when "200 OK" is received from the terminal 102, the Media information of the VoIP terminal 102 is associated with M2, and the SDP information is rewritten to M1 to send "200 OK" to the IP-PBX 103. When "ACK" is received, the Media information of the terminal 102 is associated with M1, so that the RTP is repeated via the RTP repeating installation 104. <P>COPYRIGHT: (C)2010,JPO&INPIT

Description

  The present invention relates to a SIP telephone exchange system and a SIP telephone exchange method.

  In recent years, the use of so-called IP telephones using VoIP (Voice Over IP) technology has increased. As a call control method of this VoIP system, a SIP (Session Initiation Protocol) method is known, and it is mainstream to perform call setting in accordance with IETF (Internet Engineering Taskforce) RFC3261 in which the SIP protocol is standardized.

  A procedure for call setting in the SIP system will be briefly described with reference to FIGS. Terminals 101 and 102 shown in FIG. 4 are VoIP terminals compatible with the SIP protocol, and a SIP server 303 is a call control server for processing SIP messages. Although not shown, a location server or the like necessary for establishing a session using the SIP protocol is also incorporated in the SIP server 303 or installed on a separate network.

  Now, a case where the VoIP terminal 101 makes a call to the VoIP terminal 102 will be described. The VoIP terminal 101 transmits an incoming call destination terminal ID (for example, a telephone number, here 102) to the SIP server 303 together with an “INVITE” message. At this time, the “INVITE” message from the VoIP terminal 101 is transmitted including the connection source information. For example, Media information M101 (actually a combination of an IP address and a port number, Codec, Payload) for calling the VoIP terminal 101 is set and transmitted as SDP (Service Delivery Platform) information as connection source information.

  Specifically, data “INVITE 102 (SDP information M101)” is sent to the SIP server 303 (S31 in FIG. 5). The SIP server 303 that has received “INVITE” from the VoIP terminal 101 determines that the destination is the VoIP terminal 102 from the terminal ID of the called party, and relays “INVITE”. At this time, the SDP information is also relayed as it is (S32 in FIG. 5).

  The VoIP terminal 102 that has received “INVITE” from the SIP server 303 transmits “100 Trying” and “180 Ringing”, which are provisional responses, to the SIP server 303. The SIP server 303 that has received the provisional responses “100 Trying” and “180 Ringing” relays the provisional responses to the VoIP terminal 101 that is the source (S33 in FIG. 5).

  Next, when the VoIP terminal 102 responds, a response signal “200 OK” is transmitted to the SIP server 303. At this time, in the “200 OK” from the VoIP terminal 102, Media information M102 for the VoIP terminal 102 to make a call is set as SDP information and transmitted as connection destination information. Specifically, data “200 OK (SDP information M102)” is transmitted to the SIP server 303 (S34 in FIG. 5). The SIP server 303 that has received “200 OK” relays the response signal to the VoIP terminal 101. At this time, the SDP information is also relayed as it is (S35 in FIG. 5).

  The VoIP terminal 101 that has received “200 OK” transmits “ACK” to the SIP server 303, and the SIP server 303 relays “ACK” to the VoIP terminal 102 (S36 in FIG. 5). At this point, the VoIP terminal 101 and the VoIP terminal 102 have exchanged the media information of the other party. That is, the VoIP terminal 102 exchanges the media information M101 of the VoIP terminal 101 with “INVITE”, and the VoIP terminal 101 exchanges the media information M102 of the VoIP terminal 102 with “200 OK”. As a result, a session is established between the VoIP terminal 101 and the VoIP terminal 102 which are End-End, and RTP is exchanged between the VoIP terminal 101 and the VoIP terminal 102.

  In the SIP call connection procedure described above, the media information of the calling VoIP terminal is generally included in “INVITE” as SDP information for transmission / reception. In RFC3261 where the SIP protocol is standardized, it is not mandatory to include SDP information in "INVITE", but existing SIP terminals and SIP-GW are devices that require SDP information in "INVITE". This is because there are many.

  However, in some SIP-compatible IP-PBXs and the like, there are devices that cannot add SDP information to “INVITE” due to the difference in media notification timing with the PBX side terminal. Such an IP-PBX has a problem that only a SIP terminal or SIP-GW that does not require SDP information to "INVITE" can be connected, and general-purpose SIP equipment cannot be connected by a multi-vendor or the like.

  In order to solve such a problem, the SIP control device includes an SDP description document in which the SIP message received from the terminal indicates a connection request, and information on the connection source necessary for connection is described. If it is determined that it is not, generate an SDP description document with the alternative device (SIP control device) as the connection source, add it to the SIP message and send it to the connection destination terminal. The alternative device and the connection destination terminal A SIP control technique is known in which a real data is directly transmitted and received between a terminal that is an original connection source and a terminal that is a connection destination after connection is established between For example, see Patent Document 1).

International Publication No. 2007/123096 Pamphlet (Pages 17-19, Figure 8)

  However, since the connection destination is changed after the connection is established in the technique described in Patent Document 1, there is a problem in that sound interruption occurs between the connection establishment and the connection destination change.

  This is because the SDP description document with the alternative device as the voice data reception destination is sent from the connection destination terminal to the SIP controller with the SDP description document attached to the OK response addressed to the connection source terminal, and the connection that received this transfer. The source terminal attaches an SDP description document to the ACK request addressed to the connection destination terminal and sends it to the SIP control device, and the SIP control device establishes the connection by transferring the ACK request with the SDP description document deleted to the connection destination terminal. The SIP control device transmits an SDP description document in which the connection source is transferred to the connection source terminal to the connection destination terminal in response to the voice data from the connection destination terminal, and the connection destination terminal transmits the voice data to the connection source terminal. It is because it has changed to.

  In addition, it seems that there is a possibility that 183 Session Progress, which is an Eary Media that establishes a session before 2000K, cannot be supported (silenced).

  Accordingly, an object of the present invention is to provide a SIP telephone exchange system and a SIP telephone exchange method that can connect a SIP device for general use even in an IP-PBX in which SDP information cannot be added to "INVITE" without interruption of sound. There is.

  The IP telephone exchange system according to the present invention is a call between a VoIP terminal accommodated on the PBX network side of the IP-PBX and a VoIP terminal accommodated on the SIP network side of the IP-PBX in an IP telephone exchange system using SIP. A function to relay the RTP of the time, a function to manage the VoIP terminal and Media information consisting of its own IP address and port number, and when receiving "INVITE" without SDP information from the calling VoIP terminal, Notification is made based on the function to uniquely determine the media information on the called side, set SDP information, and send “INVITE” with SDP information to the called VoIP terminal, and “200 OK” for sending “INVITE”. An RTP relay device having a function of relaying RTP by associating Media information of a VoIP terminal with the determined Media information is provided on the SIP network side.

  When the IP-PBX receives the "200 OK" transfer from the called VoIP terminal by the RTP relay device, the calling-side Media information is set to the calling-side VoIP terminal Media information, and the called-side Media information is set to "200". Media exchange as Media information added to “OK” is performed with the originating VoIP terminal, and “ACK” in which the Media information of the originating VoIP terminal is set as SDP information is transmitted to the RTP relay device.

  In addition, the IP telephone exchange method of the present invention, in the IP telephone exchange method using SIP, when the "INVITE" without SDP information is received from the calling side VoIP terminal, the media information of the calling side and the called side is uniquely determined. Decide and set SDP information, send "INVITE" with SDP information to the destination VoIP terminal, and Media information added to "200 OK" from the destination VoIP terminal for sending "INVITE" To the callee media information determined, to send “200 OK” with the determined callee media information added via IP-PBX to the caller VoIP terminal, Media exchange is performed between the IP-PBX and the originating VoIP terminal, with the media information of the originating VoIP terminal and the media information of the terminating side added to "200 OK", and the originating VoIP terminal Of sending “ACK” with the added Media information as SDP information to the RTP relay device, and the Media information added to “ACK” A step of associating the calling side Media information determined, and having a step of relaying the RTP between the originating VoIP terminal and the callee VoIP terminal by associating two Media Information.

  In the present invention, by arranging an RTP relay device at the connection destination of IP-PBX, media information is uniquely determined even before the exchange of media between terminals, and SDP information is added to "INVITE" It is characterized in that the RTP flow at the time is relayed through the transmitting terminal—the RTP relay device—the receiving terminal and the RTP relay device.

  The first effect of the present invention is that it is possible to add SDP information to “INVITE” using Media information managed by the RTP relay device through the RTP relay device. This means that even an IP-PBX that cannot add SDP information can be connected to a general-purpose SIP device.

  Further, the second effect of the present invention is that there is no interruption in the sound of a call because the addition of SDP information is performed by the RTP relay device.

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

[Description of configuration]
Referring to FIG. 1, a first embodiment of the IP telephone switching system of the present invention is shown. In FIG. 1, IP-PBX 103 is an IP-PBX that supports SIP, but SDP information is not added to “INVITE”, and media information is exchanged by “200 OK” and “ACK”. -PBX. The IP-PBX may be a SIP server.

  The VoIP terminal 101 is accommodated on the PBX network 1 side of the IP-PBX 103. It is assumed that “M101” is used as media information for a call. The VoIP terminal 102 is a terminal that requires SDP information for “INVITE”, is accommodated on the SIP network 2 side of the IP-PBX 103, and is registered with the IP-PBX 103. Assume that Media information “M102” is used for a call.

  The IP-PBX 103 transmits “INVITE” to the RTP relay device 104 when there is an incoming call to the SIP network 2 side. The incoming call to the SIP network 2 side is a case where the incoming call arrives at the VoIP terminal 102 in the configuration of FIG. Further, upon receiving “200 OK” from the RTP relay apparatus 104, the IP-PBX 103 exchanges Media information with the VoIP terminal 101 based on the received SDP information “M1”. The IP-PBX 103 that has completed the exchange of Media information with the VoIP terminal 101 sets the Media information “M101” of the VoIP terminal 101 as SDP information, and transmits “ACK” to the RTP relay apparatus 104.

  The RTP relay device 104 manages Media information (M1, M2) by itself, and has a function of relaying RTP by establishing a session based on the Media information. The RTP relay device 104 has a function of editing SDP information of the SIP message. The RTP relay device 104 is registered in the IP-PBX 103. This registration method may be station data or registration by registration.

  The RTP relay device 104 that has received “INVITE” from the VoIP terminal 101 determines two Media information (M1, M2) to be used on the calling side and the called side from the managed Media information, and receives the called Media information “ SDP information is added to “INVITE” received based on “M2” and transmitted to the VoIP terminal 102.

  The RTP relay apparatus 104 that has received “200 OK” from the VoIP terminal 102 extracts “M102” from the SDP information and associates it with the previously determined destination media information “M2”. Next, the RTP relay device 104 rewrites the media information “M102” included in the received “200 OK” SDP information with the previously determined source media information “M1” and transmits it to the IP-PBX 103.

The RTP relay apparatus 104 that has received “ACK” from the VoIP terminal 102 extracts Media information “M101” from the SDP information, and associates it with the previously determined originating Media information “M1”.
[Description of operation]
Next, the operation of the IP telephone system configured as described above will be described with reference to the sequence diagram shown in FIG.

  The IP-PBX 103 that has received the call setting request from the VoIP terminal 101 generates “INVITE”. At this time, the SDP information is not added to “INVITE” because the media information is not exchanged in the terminal 101. Since the destination is the VoIP terminal 102 of the SIP network 2, the IP-PBX 103 transmits the generated “INVITE” to the RTP relay device 104 (S11).

  The RTP relay apparatus 104 that has received “INVITE” determines two pieces of Media information (M1, M2) to be used on the calling side and the called side from the managed Media information (S12). The media information determined here is an IP address (IP address and port number of the RTP relay device 104) for receiving a TP (Trasport Protocol).

  The RTP relay device 104 adds the SDP information to the “INVITE” received based on the media information “M2” on the called side determined in S12, and transmits it to the VoIP terminal 102 (S13). The VoIP terminal 102 that has received “INVITE” returns provisional responses of “100 Trying” and “180 Ringing” and then returns “200 OK” to the RTP relay device 104 when the VoIP terminal 102 responds. . This “200 OK” includes “M102”, which is Media information of the VoIP terminal 102, in the SDP information (S14). Media information “M102” is generally the IP address and port number of the VoIP terminal 102.

  The RTP relay apparatus 104 that has received “200 OK” extracts “M102” from the SDP information and associates it with the called-side Media information “M2” determined in S12 (S15). Next, the RTP relay device 104 rewrites the media information “M102” included in the received “200 OK” SDP information with the originating side media information “M1” determined in S12, and transmits it to the IP-PBX 103 (S16). ).

  The IP-PBX 103 exchanges Media information with the VoIP terminal 101 based on the received SDP information “M1” of “200 OK” (S17). The IP-PBX 103 that has completed the exchange of Media information with the VoIP terminal 101 sets the Media information “M101” of the VoIP terminal 101 as SDP information, and transmits “ACK” to the RTP relay apparatus 104 (S18).

  The RTP relay apparatus 104 that has received “ACK” extracts the media information “M101” from the SDP information, and associates it with the originating side media information “M1” determined in S12 (S19). By following the procedure as described above, the terminal 101 transmits RTP to the destination Media information “M1”, and the VoIP terminal 102 transmits RTP to the destination Media information “M2”.

  Since the IP addresses actually set in the media information “M1” and “M2” are the IP addresses of the RTP relay device 104, both the terminals 101 and 102 transmit RTP to the RTP relay device 104. The RTP relay device 104 establishes a session between the media information “M1” and the media information “M2” determined in S2, and relays the RTP.

  Specifically, the RTP relayed to the media information “M1” is relayed to “M102” associated in S15. That is, VoIP terminal 101 → M1 → M2 → M102 (VoIP terminal 102) and RTP are transmitted. Further, the RTP relayed to the media information “M2” is relayed to “M101” associated in S19. That is, RTP is transmitted as VoIP terminal 102 → M2 → M1 → M101 (VoIP terminal 101).

  In this way, since it is possible to add SDP information to “INVITE” via the RTP relay device 104, even if the IP-PBX 103 cannot add SDP information to “INVITE”, “INVITE” It is possible to connect to a terminal 103 that requires SDP information.

  In the present invention, since the media information “M1” and the media information “M2” of the RTP relay device 104 that actually exchanges audio data are added as SDP information to establish a connection, sound interruption or the like may occur. Absent. In addition, regarding Eary Media, it is possible to avoid silence by a method such as conversion to 180 Ringing by the RTP relay device 104.

  Furthermore, in this embodiment, the RTP relay device 104 is described as a device that only relays RTP. However, if the RTP relay device 104 is positioned as a device having a mechanism that can also perform media conversion internally, it depends on each terminal or network. Even if Codec and Payload are different, connection is possible without problems.

  Moreover, since the call that was sending and receiving RTP at End-End goes through RTP relay device 104, even if the end-end terminal does not have a recording function, the call is recorded by RTP relay device 104. You can also use it.

  In the above embodiment, the RTP relay device 104 is a single device, but may be a single module in the IP-PBX 103.

  SIP telephone exchange that can be applied even when receiving "INVITE" with SDP information added from VoIP terminal 101 by adding a function to determine the presence or absence of SDP information to IP-PBX103 and RTP relay device 104 It can be a system. FIG. 3 shows a sequence diagram when SDP information is added to “INVITE”.

  The SIP server 203 in this SIP telephone switching system has the same function as the IP-PBX 103 when it receives a call setting request to which no SDP information is added from the VoIP terminal 101. On the other hand, when SDP information including Media information “M101” of VoIP terminal 101 is added to “INVITE” received from VoIP terminal 101, the SDP information is directly relayed to RTP relay apparatus 104 (FIG. 3). S21). Also, there is no Media exchange (S17) in FIG. 2, and SDP information “M1” of “200 OK” received from the RTP relay device 204 (S27 in FIG. 3) is directly transferred to the VoIP terminal 101 (S28 in FIG. 3). . After this, “ACK” flows as follows: VoIP terminal 101 → IP-SIP server 203 → RTP relay device 104 → VoIP terminal 102, and a call that relays RTP relay device 204 between VoIP terminal 101 and VoIP terminal 102 Done.

  Further, the RTP relay device 204 in this SIP telephone exchange system has the same function as the RTP relay device 104 when the SDP information is not added to the “INVITE” received from the SIP server 203. On the other hand, when SDP information is added, two Media information (M1, M2) to be used on the calling side and the called side are determined from the managed Media information (S22 in FIG. 3). As in S19 of 2, the media information “M101” is extracted from the SDP information, and is associated with the determined media information “M1” of the originating side (S23). Similar to S13 of FIG. SDP information is added to “INVITE” received based on Media information “M2” and transmitted to VoIP terminal 102 (S24).

  The remaining processes S25 and S26 are the same as S14 and S15 in FIG.

Block diagram of VoIP telephone system of the present invention Sequence diagram showing the operation of the VoIP telephone system shown in FIG. Sequence diagram showing one operation of another VoIP telephone system of the present invention Block diagram showing an example of a conventional VoIP telephone system Sequence diagram showing the operation of the VoIP telephone system shown in FIG.

Explanation of symbols

1 PBX network 2 SIP network
101,102 VoIP terminal
103 IP-PBX
104 RTP relay equipment
203 SIP server

Claims (6)

In an IP telephone switching system using SIP,
A function of relaying RTP during a call between a VoIP terminal accommodated on the PBX network side of the IP-PBX and a VoIP terminal accommodated on the SIP network side of the IP-PBX;
A function for managing Media information including the VoIP terminal and its own IP address and port number;
When “INVITE” without SDP information is received from the calling VoIP terminal, the media information of the calling and called parties is uniquely determined and SDP information is set, and “INVITE” with SDP information is set to the called VoIP terminal. With the ability to send to
An RTP relay device having a function of relaying the RTP by associating the Media information of the VoIP terminal notified based on “200 OK” for the transmission of the “INVITE” and the determined Media information on the SIP network side An IP telephone switching system characterized by its deployment.   When the IP-PBX receives the “200 OK” transfer from the destination VoIP terminal by the RTP relay device, the IP-PBX displays the media information of the calling side VoIP terminal, the media information of the calling side VoIP terminal, and the media information of the destination side. Media exchange as Media information added to the “200 OK” is performed with the originating VoIP terminal, and “ACK” in which the Media information of the originating VoIP terminal is set as SDP information is used as the RTP relay device The IP telephone switching system according to claim 1, wherein the IP telephone switching system transmits to the IP telephone.   The IP telephone exchange system according to claim 1 or 2, wherein a function for determining the presence or absence of SDP information is added to the IP-PBX 103 and the RTP relay device 104.   4. The IP telephone exchange system according to claim 1, wherein the IP-PBX is replaced with a SIP server.   The IP telephone exchange system according to claim 1, wherein the RTP relay device has a recording function for recording RTP to be relayed. In the IP telephone exchange method using SIP,
When “INVITE” without SDP information is received from the calling VoIP terminal, the media information of the calling and called parties is uniquely determined and SDP information is set, and “INVITE” with SDP information is set to the called VoIP terminal. Sending to
Associating the media information added to "200 OK" from the destination VoIP terminal with respect to the transmission of the "INVITE" and the determined destination media information;
Sending `` 200 OK '' to which the determined calling side Media information is added to the calling side VoIP terminal via the IP-PBX;
Media exchange between the IP-PBX and the originating VoIP terminal is performed by using the originating Media information as the Media information of the originating VoIP terminal and the terminating Media information as the Media information added to the “200 OK”. Transmitting `` ACK '' with the media information of the calling VoIP terminal added as SDP information to the RTP relay device,
Associating the media information added to the "ACK" with the determined originating media information;
An IP telephone exchange method comprising: relaying the RTP between the calling VoIP terminal and the called VoIP terminal by associating the two Media information.

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