JP4795027B2 - Communication apparatus and communication system - Google Patents

Description

The present invention relates to a communication apparatus and a communication system having a short packet multiplexing / separating function, and in particular, in VoIP (Voice Over Internet Protocol), a plurality of short packets containing voice data are multiplexed into one IP packet. The present invention relates to a communication device and a communication system.

As an IP (Internet Protocol) -based communication apparatus having a short packet multiplexing / demultiplexing function, for example, an IP-based communication apparatus shown in ITU-T (International Telecommunication Union-Telecommunication standard) G.769 as Non-Patent Document 1 is used. There is a circuit multiplexing equipment (IPCME). This recommendation proposes a method of multiplexing and transmitting a plurality of short packets in RTP (Real-time Transpoprt Protocol) / UDP (Ueser Datagram Protocol) / IP packets. The IP packet shown in FIG. 2 to be described later is a method in accordance with this recommendation, and a plurality of short packets are multiplexed in the RTP payload using a VoIP (Voice Over Internet Protocol) standard protocol called RTP / UDP / IP. To do.

At present, as a VoIP call control system, SIP (Session Initiation Protocol) standardized by RFC (Request For Comments) 3261 to 3264 of IETF (Internet Engineering Task Force) is mainly used. SIP is a text-based protocol similar to HTTP (HyperText Transfer Protocol), and a SIP message is composed of a SIP header group and a SIP body. The SIP body uses a description rule called SDP (Session Description Protocol), describes voice session information (for example, encoding method and port number), and is used for end-to-end negotiation.
Audio data is transmitted using RTP standardized by RFC3550 or the like.

  In a network that operates an IP telephone using the SIP method, an IP packet carrying an SIP message and an IP packet carrying an RTP are transmitted through different sessions and routes, respectively. Note that UDP is generally used as the transport layer protocol of SIP and RTP, but TCP (Transmission Control Protocol) / IP is also possible.

  In a VoIP network based on SIP, an IP telephone server called a SIP proxy is installed, and a VoIP terminal such as an IP telephone stores the SIP proxy. The positioning is the same as the DNS (Domain Name System) in HTTP. This SIP proxy is a circuit switching server for an IP phone that performs location registration of the IP phone and control of incoming / outgoing / incoming / disconnected.

  On the other hand, a media proxy (or RTP proxy) is a proxy for voice data and image data, and is a device that is installed at the boundary of a network in the same way as an HTTP proxy, and once aggregates and transmits channels of a plurality of terminals. Used as a gateway or NAT (Network Address Translator) / firewall transit means.

  When the SIP proxy and media proxy are integrated, it becomes a server having the functions of the SIP proxy and media proxy, and it behaves as two IP phone terminals back to back, and performs SIP proxy processing and media proxy processing, so B2BUA (Back to Back User Agent).

  SIP is a text-based communication procedure similar to HTTP, and it is easier to assemble and expand the communication system than other methods, so that the communication system can be realized with high efficiency and low cost.

  Further, as a conventional communication device or communication system, for example, in Patent Document 1, when transmitting information signals of a plurality of calls with the same destination network connection device, it is configured and transmitted as an IP packet with less overhead in communication. In Japanese Patent Application Laid-Open No. 2004-228688, the quality of the video / audio signal reproduced from the content is not deteriorated on the side where the content to be distributed is acquired, and higher concealment is achieved in a shorter time. A content distribution system capable of transmitting video / audio signals of content while maintaining the performance is disclosed.

Japanese Patent No. 3319367 (paragraph 0006) JP 2004-96648 A (paragraph 0014) ITU-T G.769, Aug.2002

  The conventional communication apparatus and communication system are configured as described above. For example, the IP-based line multiplexing apparatus (IPCME) shown in Non-Patent Document 1 is a VoIP call control method that is mainstream today. It is not compatible with SIP, so it cannot be applied to existing VoIP systems. It can reduce the contracted bandwidth of access lines in a corporate communication network when trying to realize VoIP in a limited band such as a satellite line. When trying to realize QoS (Quality of Service) of a voice line, there is a problem that it cannot be realized with high efficiency and low cost.

  The present invention has been made to solve the above-described problems, and is intended to realize VoIP in a limited band such as a satellite line, or to reduce the contracted bandwidth of an access line in an enterprise communication network or a voice line. An object of the present invention is to obtain a communication device and a communication system that can be realized with high efficiency and low cost by applying SIP when trying to realize QoS.

The communication apparatus according to the present invention registers a local IPCME domain indicating a multiplexing target area of a local station and a remote IPCME domain indicating a multiplexing target area of a partner station when multiplexing voice data, and a transmission source of a received SIP message is When the local IPCME domain of the local station matches and the destination of the received SIP message matches the remote IPCME domain of the partner station, the received SIP message is transferred to the partner station, and the SIP message received from the partner station is transmitted. When the destination matches the local IPCME domain of the local station, the received SIP message is transferred to the transmission destination of the local station and the SIP message is received when the SIP message is received from the transmission source of the local IPCME domain of the local station. In the local SIPCME domain of the local station When the IPCME-Gateway header describing the text data to be included is added and the SIP message is received from the remote IPCME domain sender of the partner station, the partner station described in the IPCME-Gateway header added to the SIP header is added. A SIP proxy unit that performs call connection control by acquiring and registering a remote IPCME domain, and an IPCME packet based on call connection control by the SIP proxy unit by multiplexing RTP packets of voice data from the transmission source of the local station And a media proxy unit that separates the IPCME packet received from the partner station and forwards the RTP packet to the transmission destination of the own station .

  According to the present invention, a communication device can be operated coexisting with an existing SIP terminal, and an IP terminal or an end user can use a voice line seamlessly without being aware of the presence of the communication device. Specifically, there is an effect that a communication carrier can benefit from high-efficiency coded multiplex transmission and can efficiently operate a voice line at low cost.

Embodiment 1 FIG.
1 is a block diagram showing a configuration of a communication system according to Embodiment 1 of the present invention. In this communication system, a center station 1, a very small ground station (VSAT) 2 and a very small ground station 3 communicate via an artificial satellite 4, and the very small ground station 2 and the very small ground station 3 are connected. Performs communication by the double hop relay via the artificial satellite 4 and the center station 1. The center station 1 includes a line multiplexing device (IPCME) 11a, an IP telephone (or terminal adapter) 12a, an IP-PBX (Interenet Protocol-Private Branch eXchange) 13a, a satellite gateway device (Sat-GW) 14a, and an Ethernet (registered trademark). ) A cable 15 a is provided, and the IP-PBX 13 a is connected to the public line network 5.

The micro ground station 2 includes a line multiplexing device (IPCME) 11b, an IP telephone (or terminal adapter) 12b, an IP-PBX 13b, a satellite gateway device (Sat-GW) 14b, and an Ethernet (registered trademark) cable 15b. The IP-PBX 13b is connected to the public network 6. The micro ground station 3 includes a line multiplexing device (IPCME) 11c, an IP telephone (or terminal adapter) 12c, a satellite gateway device (Sat-GW) 14c, and an Ethernet (registered trademark) cable 15c.

In the center station 1, the micro ground station 2, and the micro ground station 3, the satellite gateway devices 14a, 14b, and 14c communicate with each other via the artificial satellite 4, and the line multiplexers 11a, 11b, and 11c transmit the SIP message. SIP proxy function for terminating / editing / forwarding, media proxy function for terminating / editing / forwarding voice data, and short packet multiplexing / separating function for multiplexing or separating short packets. IP telephones 12a, 12b, 12c and IP-PBXs 13a, 13b are SIP terminals adopting the SIP system, and Ethernet (registered trademark) cables 15a, 15b, 15c connect each device in each station.
In this specification and FIG. 1, network devices such as routers and switching hubs are omitted.

  As described above, since the line multiplexers 11a, 11b, and 11c have the short packet multiplexing / separating function, it is possible to reduce the overhead and the number of packets by multiplexing, and to reduce the bandwidth of the voice line. It can be greatly reduced. Therefore, it can be effectively applied to satellite communication, communication between bases, and remote island communication.

  The line multiplexers 11a, 11b, and 11c installed in the center station 1, the micro ground station 2, and the micro ground station 3 are communication devices that integrate the SIP proxy function, the media proxy function, and the short packet multiplexing / demultiplexing function. The communication device may be provided in the satellite gateway devices 14a, 14b, and 14c installed in the center station 1, the micro ground station 2, and the micro ground station 3.

  FIG. 2 is a diagram showing a VoIP packet used in the communication system according to Embodiment 1 of the present invention. In the center station 1, the micro ground station 2, and the micro ground station 3, a normal VoIP packet (RTP packet) between the line multiplexer 11 and the IP telephone 12 is voice data encoded with the header unit 100. The header part 100 is composed of an IP header 101, a UDP header 102, and an RTP header 103.

A VoIP packet (IPCME packet) between the line multiplexers 11 is composed of a header part 100 and short packets 105-1, 105-2, ..., 105-x. The header part 100 has an IP header 101 and a UDP header. 102 and the RTP header 103. Each short packet 105-1, 105-2,..., 105-x is composed of a header part 106 and a payload part 107 which is voice-coded voice data. Yes. The short packets 105-1, 105-2,..., 105-x can accommodate voice data for a plurality of channels in one IP packet.
Here, the channel is one call and consent, but when multiplexing, a plurality of data on the same channel may be multiplexed into one packet.

  2 shows a VoIP packet used by the media proxy unit 22 in a call connection sequence described later, and FIG. 2 does not show a packet used by the SIP proxy unit 21 in a call connection sequence described later.

  FIG. 3 is a diagram showing an example of a call connection sequence between the micro ground station 2 and the center station 1 in the communication system according to Embodiment 1 of the present invention. Here, it is assumed that the call connection between the IP phones 12b and 12a is performed. In FIG. 3, IPCME domains 10a and 10b indicate areas to be multiplexed when the line multiplexers 11a and 11b multiplex voice data. For example, a domain expression such as a.ipcme.com or 192.168.1.0/ Network representation such as 24 is used, and is usually set for each station of the center station 1, the micro ground station 2, and the micro ground station 3.

  Although not shown in FIG. 3, the line multiplexer 11c of the micro ground station 3 also has the IPCME domain 10c. The IPCME domains 10a, 10b, and 10c are registered in the line multiplexers 11a, 11b, and 11c.

  The line multiplexers 11a and 11b include SIP proxy units 21a and 21b and media proxy units 22a and 22b, which are internal function entities, respectively. Although not shown in FIG. 3, the line multiplexer 11c of the micro ground station 3 similarly includes the SIP proxy unit 21c and the media proxy unit 22c.

  The SIP proxy unit 21 registers the local IPCME domain indicating the multiplexing target area of the local station and the remote IPCME domain indicating the multiplexing target area of the partner station when the voice data is multiplexed, and the transmission source of the received SIP message is the local station When the destination of the received SIP message matches the remote IPCME domain of the other station, the call connection control is performed by transferring the received SIP message to the other station and received from the other station. When the transmission destination of the SIP message matches the local IPCME domain of the local station, the received SIP message is transferred to the transmission destination of the local station.

  Based on the call connection control by the SIP proxy unit 21, the media proxy unit 22 multiplexes the RTP packet of the voice data from the transmission source of the local station, transfers the IPCME packet to the partner station, and receives the IPCME packet received from the partner station. The RTP packet is separated and transferred to the transmission destination of the local station.

  In FIG. 3, F1, F2, and F3 are flows of INVITE messages (one of SIP messages) that are issued when the IP phone 12b of the micro ground station 2 is off-hooked and a call is made. F10, F11, F12 Is a flow of a 200 OK message (one of the SIP messages, the 200 series being a successful response) issued when the IP phone 12a of the center station 1 picks up the handset. F16, F18, F19, and F21 are voice data flows using normal VoIP packets (RTP packets), and F17 and F20 are voice data multiplex transmission flows using VoIP packets (IPCME packets) between the line multiplexers 11b and 11a. .

  F11s is a flow for transmitting operation information from the SIP proxy unit 21a to the media proxy unit 22a within the line multiplexer 11a simultaneously with the flow of F11. F12s is a flow for transmitting operation information from the SIP proxy unit 21b to the media proxy unit 22b within the line multiplexing device 11b simultaneously with the flow of F12.

  The operation information transmitted by F11s and F12s is the IPCME standby port number in F17 and F20, the RTP standby port number in F16 and F19, and the channel ID. The IPCME standby port number can be determined in advance, and in this case, transmission is not necessary. The channel ID is an ID for identifying a call in the multiplex flows F17 and F20 of voice data by the line multiplexers 11b and 11a. From the Call-ID transmitted by SIP, the channel ID is between the line multiplexers 11b and 11a. It is associated with the short packet header value transmitted in the VoIP packet. A plurality of calls between the same end and end can be identified by this channel ID.

FIG. 4 is a diagram showing an example of a call connection sequence between the micro ground station 2 and the center station 1 in the communication system according to Embodiment 1 of the present invention, and shows details of the call connection sequence of FIG.
F1, F2, and F3 are flows of an INVITE message that is issued when an IP phone 12b is off-hooked and a call is made. F4, F5, and F6 are 100 Trying messages (one of SIP messages) indicating that call connection processing is in progress. , 100 series is a provisional response) flow. F7, F8, and F9 are flows of a 180 Ringing message (one of SIP messages) indicating that the other party is being called, and F10, F11, and F12 are flows of a 200 OK message indicating that the call connection has been accepted.

  F13, F14, and F15 are flows of ACK messages (one of SIP messages), F16 to F21 are flows of voice data, and F16, F18, F19, and F21 are flows of transmission of normal VoIP packets (RTP packets). F17 and F20 are multiplex transmission flows of voice data by VoIP packets (IPCME packets) between the line multiplexers 11a and 11b. F22 to F27 are call disconnection flows using a BYE message (one of SIP messages) and a 200 OK message.

In FIG. 4, the flow of F2, F4, F9, F12, F14, F23, and F27 is the processing of the SIP proxy unit 21b of the line multiplexer 11b, and the flows of F3, F5, F8, F11, F15, F24, and F26 are In the processing of the SIP proxy unit 21a of the line multiplexer 11a, the processing of F17 and F21 is the processing of the media proxy unit 22b, and the processing of F18 and F20 is the processing of the media proxy unit 22a.
Note that the sequence shown in FIG. 4 is simplified for the sake of explanation, and may not actually be this way.

5 and 6 are flowcharts showing an example of processing of the INVITE message of the SIP proxy unit 21 of the line multiplexer 11 in the communication system according to Embodiment 1 of the present invention.
As a premise, it is assumed that the IPCME domain 10 indicating a multiplexing target area when multiplexing audio data is registered in the line multiplexing apparatus 11 in advance. The domain of the local station is called a local IPCME domain, and the domain of the counterpart station is called a remote IPCME domain. For example, when viewed from the line multiplexer 11b of the micro ground station 2, the IPCME domain 10b is a local IPCME domain, and the IPCME domains 10a and 10c are remote IPCME domains.
Here, the processing unique to the line multiplexing apparatus 11 other than the standard SIP processing based on RFC3261 will be mainly described along this flowchart.

  In FIG. 5, when the SIP proxy unit 21 of the line multiplexing device 11 receives the INVITE message, in step ST11, the SIP proxy unit 21 matches the line multiplexing device 11 of the partner station whose source IP address is registered in advance. Check whether or not to do. If they match, it is an INVITE message issued from the line multiplexer 11 of the partner station, so the process proceeds to step ST21 in FIG. 6. If they do not match, for example, IP telephone 12 or IP- Since this is an INVITE message from the SIP terminal of PBX 13, the process proceeds to step ST12.

  In step ST12, the SIP proxy unit 21 checks whether the destination of the SIP-URI (SIP-Uniform Resource Identifier) or the To header hits a previously registered remote IPCME domain.

If there is no hit in step ST12, it is not an INVITE message addressed to the line multiplexing apparatus 11 managing the remote IPCME domain, so the process proceeds to standard SIP processing based on RFC3261 in step ST18. If the transfer rule is hit, the INVITE message is transferred. If no transfer rule is hit, a 404 Not Found message (400 is a request error response) is returned. Here, the transfer rule indicates, for example, routing according to Record-Route of the SIP header, or routing set in advance by the SIP proxy unit 21 based on the SIP-URI.
If there is a hit in step ST12, it is an INVITE message addressed to the line multiplexing apparatus 11 managing the remote IPCME domain, so the process proceeds to step ST13.

In step ST13, the SIP proxy unit 21 checks whether the terminal or source IP address indicated by the From header hits a pre-registered local IPCME domain.
If there is no hit, the target is not multiplexed, so the process proceeds to the standard SIP process based on RFC3261 in step ST18. If it is hit, the target is multiplexed, and the process proceeds to step ST14.

In step ST14, the SIP proxy unit 21 checks whether or not the status information of the destination line multiplexer 11 is normal. Here, the status information includes the surviving information (keep alive) of the destination line multiplexer 11, the presence / absence of error information of the destination line multiplexer 11, the bandwidth limit over with the partner line multiplexer 11, For example, the channel number limit with the line multiplexer 11 is exceeded.
If the status information is abnormal, the process proceeds to step ST19, a BUSY message response is returned to the transmission source, the call connection is rejected, and the SIP processing related to reception of the INVITE message is terminated. If the status information is normal, the process proceeds to step ST15.

In step ST15, the SIP proxy unit 21 adds an IPCME-Gateway header to the SIP header. The IPCME-Gateway header added here is composed of the following text data.
IPCME-Gateway: <IP address of this line multiplexer 11><Portnumber><ID><IPCME domain (option)>
Here, the port number is the standby port number of the line multiplexing apparatus 11, but is not necessary if it is determined in advance. The ID corresponds to a Call-ID that uniquely represents a call in SIP, and is associated with a short packet header. The IPCME domain is for automatically adding an IPCME domain to the other party, but is not necessary if the remote IPCME is registered in advance in each line multiplexer 11.
Note that FIGS. 7 and 10 to be described later show a specific expression example of the IPCME-Gateway header.

  After completing the process of step ST15, in step ST16, the SIP proxy unit 21 stores the Call-ID in the IPCME session management area provided in the SIP proxy unit 21. This storage area is referred to when 200OK is received. Finally, in step ST17, the SIP proxy unit 21 performs standard SIP processing based on RFC3261, and after processing the SIP header, as shown in the flow F2 in FIGS. 3 and 4, the line multiplexing for managing the remote IPCME domain The INVITE message is transferred to the conversion apparatus 11. Here, as processing of the SIP header, a line describing the IP address is added to the Record-Route header, or the SIP-URI is rewritten.

  In step ST13, if the terminal or source IP address indicated in the From header does not hit the local IPCME domain, the SIP proxy unit 21 performs standard SIP processing based on RFC3261 because it is not a target for multiplexing in step ST18. If the transfer rule is hit, the INVITE message is transferred, and if not hit, a 404 Not Found response is made.

  The process of FIG. 6 is a process of the INVITE message transferred from the line multiplexer 11 that manages the remote IPCME domain. In step ST21, the SIP proxy unit 21 has an IPCME-Gateway header in the SIP header of the received INVITE message. Check whether or not. If there is no IPCME-Gateway header, the process proceeds to step ST26, where normal VoIP processing is performed, and after processing the SIP header based on RFC3261, an INVITE message is transferred to a terminal such as the IP telephone 12. If there is an IPCME-Gateway header, the process proceeds to step ST22.

  In step ST22, the SIP proxy unit 21 checks whether the destination of the received INVITE message hits the local IPCME domain. If there is no hit, the process proceeds to step ST25. If there is a hit, the process proceeds to step ST23.

In step ST23, the SIP proxy unit 21 rewrites the SDP description in the INVITE message. The rewriting target is an IP address of one-character mnemonic “o =”, an IP address of “c =”, and a port number of “m =”. Here, “o” is used to identify the session initiator and the session, “c” is used to identify the network type, address type, and session initiator. “M” is the media type, port number, and transport protocol. Used to identify media list. The rewrite contents here are the IP address of “o =” and the IP address of “c =”, the IP address of this line multiplexer 11, and the port number of the flow of F19 that this line multiplexer 11 waits for. RTP standby port number.
FIG. 8 described later is an example of the INVITE message in step ST23, and the underlined portion is the SDP rewrite portion.

  In step ST <b> 24, the SIP proxy unit 21 stores a set called a Call dialog composed of a Call-ID or a Call-ID, a From tag, and a To tag in an IPCME session management area provided in the SIP proxy unit 21.

  In step ST25, the SIP proxy unit 21 acquires the text data described in the IPCME-Gateway header, and then deletes the IPCME-Gateway header. Here, as a specific means for acquiring the text data, a method of storing it in a database or a management table on a memory can be considered. If such functions and means are secured, the network operation manager does not have to input text data in advance, and does not have to register a remote IPCME domain or the like in advance.

  Finally, in step ST26, the SIP proxy unit 21 performs standard SIP processing based on RFC3261, and transfers the INVITE message to a terminal such as the IP telephone 12 after processing the SIP header.

  FIGS. 7 and 8 are diagrams showing examples of INVITE messages output from the line multiplexer 11. 7 corresponds to the INVITE message between the line multiplexers 11 in the flow of F2 in FIGS. 3 and 4 and the flow of G2 and G5 in FIG. FIG. 8 corresponds to the INVITE message in the flow of F3 in FIGS. 3 and 4 and the flow of G3 and G6 in FIG.

  In the INVITE message shown in FIGS. 7 and 8, each line in the upper part of the blank line is a SIP header, and in FIG. 7, an IPCME-Gateway header is added. The lower part of the blank line is the SDP. In FIG. 8, the SDP single-character mnemonic “o =” IP address, “c =” IP address, and “m =” port number are rewritten. .

  FIG. 9 is a flowchart showing an example of processing of the 200 OK message of the SIP proxy unit 21 of the line multiplexer 11 in the communication system according to Embodiment 1 of the present invention. In step ST31, the SIP proxy unit 21 of the line multiplexing device 11 that has received the 200OK message checks whether the Call-ID hits the one registered in step ST16 in FIG. 5 or step ST24 in FIG. To do. If there is no hit, the process proceeds to a normal VoIP process in step ST35, the transfer process described in the Record-Route header is performed, and if a hit is found, the process proceeds to step ST32.

  In step ST32, the SIP proxy unit 21 checks whether or not there is an IPCME-Gateway header. If there is an IPCME-Gateway header, the process proceeds to step ST41 as the calling-side line multiplexer 11; if there is no IPCME-Gateway header, the process at step ST33 as the called-side line multiplexer 11 Proceed to

  In Step ST33, the SIP proxy unit 21 of the called-side line multiplexer 11 adds an IPCME-Gateway header to the SIP header. The content of the IPCME-Gateway header to be added is the same as the content of step ST15 in FIG. Thereafter, in step ST34, the SIP proxy unit 21 performs standard SIP processing based on RFC3261 and performs transfer processing described in the Record-Route header.

  In step ST41, the SIP proxy unit 21 of the line multiplexing device 11 on the calling side performs a rewriting process of the SDP description. The rewrite target is an IP address of “o =”, an IP address of “c =”, and a port number of “m =”. The rewrite contents here are the IP address of “o =” and the IP address of “c =”, the IP address of the line multiplexer 11, and the port number are waiting for this line multiplexer 11. 4 is an RTP standby port number of the flow of F16 shown in FIG. FIG. 11 described later is an example of the 200OK message in step ST41, and the underlined portion is the SDP rewrite portion.

  In step ST42, the SIP proxy unit 21 deletes the IPCME-Gateway header after acquiring the text data described in the IPCME-Gateway header, as in step ST25 of FIG. Thereafter, in step ST43, the SIP proxy unit 21 performs standard SIP processing based on RFC3261 and performs transfer processing described in the Record-Route header.

  10 and 11 are diagrams showing examples of 200 OK messages output from the line multiplexer 11. FIG. 10 corresponds to the SIP message between the line multiplexers 11 in the flow of F11 shown in FIGS. 3 and 4 and the flow of G8 and G11 in FIG. 11 corresponds to the SIP message in the flow of F12 shown in FIGS. 3 and 4 and the flow of G9 and G12 in FIG.

  In the 200 OK message shown in FIG. 10 and FIG. 11, each line in the upper part of the blank line is a SIP header, and in FIG. 10, an IPCME-Gateway header is added. Further, the lower part of the blank line is the SDP. In FIG. 11, the SDP one-character mnemonic “o =” IP address, “c =” IP address, and “m =” port number are rewritten. .

FIG. 12 is a diagram showing a call connection sequence between the micro ground station 2 and the micro ground station 3 in the communication system according to Embodiment 1 of the present invention. In this call connection sequence, communication is performed via the IP-PBX 13 a installed in the center station 1. In the center station 1, it is assumed that the IP-PBX 13a has a second call function or a B2BUA function, and the line multiplexer 11a-1 is installed for communication with the micro ground station 2, and the line multiplexer 11 a-2 is installed for communication with the micro ground station 3. The line multiplexer 11a-1 and the line multiplexer 11a-2 may be a single line multiplexer 11a.
In FIG. 12, description of SIP messages that are not directly related to the present invention, such as 100 Trying, 180 Ringing, and PRACK, is omitted. The equipment shown in FIG. 12 is the same as each equipment shown in FIG.

  Communication between the micro ground station 2 and the center station 1 shown in FIG. 12 is the same as the sequence shown in FIG. In the communication between the center station 1 and the micro ground station 3 shown in FIG. 12, the micro ground station 2 is replaced with the center station 1 and the center station 1 is replaced with the micro ground station 3 in the sequence shown in FIG. Sequence.

  Regarding the processing of each line multiplexer 11 in FIG. 12, the flow of G2, G3, G5, G6, G8, G9, G11, G12, G14, G15, G17, G18 is the processing of the SIP proxy unit 21 shown in FIG. The flow of G20, G21, G23, G24, G26, G27, G29, and G30 is the processing of the media proxy unit 22 shown in FIG.

  When the IP telephone 12b of the micro ground station 2 makes a call to the IP telephone 12c of the micro ground station 3, the IP telephone 12b transmits an INVITE message to the line multiplexer 11b in accordance with the flow of G1. The line multiplexer 11b performs the processing of steps ST11 to ST17 in the flowchart shown in FIG. 5, and transmits the INVITE message to the line multiplexer 11a-1 of the center station 1 in the flow of G2.

  The line multiplexing apparatus 11a-1 of the center station 1 performs the processing of steps ST11 and ST21 to ST26 in the flowcharts shown in FIGS. 5 and 6, and transmits an INVITE message to the IP-PBX 13a in the flow of G3. The IP-PBX 13a transmits a new INVITE message independent of the INVITE messages of the G1 to G3 flows to the line multiplexer 11a-2 by the second call function or the B2BUA function, and the micro ground station 3 Wait for the 200OK message from the side. The line multiplexer 11a-2 performs the processing of steps ST11 to ST17 in the flowchart shown in FIG. 5, and transmits the INVITE message to the line multiplexer 11c of the micro ground station 3 in the flow of G5.

  The line multiplexer 11c of the micro ground station 3 performs the processing of steps ST11, ST21 to ST26 in the flowcharts shown in FIGS. 5 and 6, and transmits an INVITE message to the IP phone 12c in the flow of G6.

  When the IP phone 12c receives the INVITE message in the flow of G6 and picks up the receiver, it transmits a 200 OK message to the line multiplexer 11c in the flow of G7. The line multiplexer 11c performs the processing of steps ST31 to ST34 in the flowchart shown in FIG. 9, and transmits a 200 OK message to the line multiplexer 11a-2 of the center station 1 in the flow of G8.

  The line multiplexer 11a-2 of the center station 1 performs the processing of steps ST31, ST32, ST41 to ST43 in the flowchart shown in FIG. 9, and transmits a 200OK message to the IP-PBX 13a in the flow of G9. In response to the completion of the call connection on the micro ground station 3 side, the IP-PBX 13a transmits a 200 OK message to the line multiplexer 11a-1 according to the flow of G10. The line multiplexer 11a-1 performs the processing of steps ST31 to ST34 in the flowchart shown in FIG. 9, and transmits a 200 OK message to the line multiplexer 11b of the micro ground station 2 in the flow of G11.

  The line multiplexer 11b of the micro ground station 2 performs the processing of steps ST31, ST32, ST41 to ST43 of the flowchart shown in FIG. 9, and transmits a 200 OK message to the IP phone 12b in the flow of G12.

  The IP telephone 12b receives the 200OK message in the G12 flow, returns an ACK message to the line multiplexing apparatus 11b in the G13 flow, and starts sending an RTP packet carrying voice data in the G19 flow. Here, the ACK message for the 200 OK message is processing in accordance with standard SIP and is transferred in the flow of G13 to G18. However, for the ACK message of the flow of G16, it is not necessary to wait for the reception of the ACK message of the flow of G15. Alternatively, it may be transmitted simultaneously with the transmission of the 200 OK message in the G10 flow.

  The line multiplexer 11b that has received the RTP packet in the flow of G19 places the voice data included in the RTP packet on the short packet, and if there are a plurality of short packets, generates a RTP packet in which these are multiplexed, The IPCME packet is transmitted to the line multiplexer 11a-1 of the center station 1 in accordance with the flow shown in FIG.

  The line multiplexer 11a-1 of the center station 1 disassembles the IPCME packet received in the G20 flow, extracts a short packet, and transmits the VoIP standard RTP packet to the IP-PBX 13a in the G21 flow. When the IP-PBX 13a extracts voice data from the RTP packet received in the G21 flow by the second call function or the B2BUA function, the line multiplexing device newly adds the VoIP standard RTP packet in the G22 flow without changing the voice data. 11a-2. The line multiplexer 11a-2 that has received the RTP packet in the G22 flow places the voice data included in the RTP packet on the short packet, and if there are a plurality of short packets, generates a RTP packet in which the short packets are multiplexed, The IPCME packet is transmitted to the line multiplexer 11c of the micro ground station 3 by the flow of G23.

The line multiplexer 11c of the micro ground station 3 disassembles the IPCME packet received in the G23 flow, extracts a short packet, and transmits the VoIP standard RTP packet to the IP phone 12c in the G24 flow. The IP phone 12c extracts voice data from the RTP packet received in the G24 flow, performs decoding processing, etc., and reproduces it to the receiver.
About the flow of G25-G30, it is the same as that of the process in the flow of said G19-G24.

  As described above, according to the first embodiment, the line multiplexing apparatus 11 of the center station 1 and the microminiature ground stations 2 and 3 includes the SIP proxy unit 21. The SIP proxy unit 21 is, for example, shown in FIGS. By performing the processing for the SIP message shown in FIG. 6, the line multiplexer 11 can be operated in coexistence with the existing SIP terminal, and the IP terminal and the end user are aware of the presence of the line multiplexer 11. Voice line can be used seamlessly, and finally, the telecommunications carrier can benefit from high-efficiency coded multiplex transmission by applying the line multiplexer 11 and operate the voice line efficiently at low cost. The effect that it can do is acquired.

  Further, according to the first embodiment, in the case where there are a plurality of micro ground stations, and the center station 1 and the micro ground station 2 are large and the frequency of IP telephones is high, the center station 1 and the super ground station 2 The effect of multiplexing between two small ground stations can be determined, and the effect of enabling high-efficiency operation is obtained.

It is a block diagram which shows the structure of the communication system by Embodiment 1 of this invention. It is a figure which shows the VoIP packet used with the communication system by Embodiment 1 of this invention. It is a figure which shows an example of the call connection sequence of a micro ground station and a center station in the communication system by Embodiment 1 of this invention. It is a figure which shows an example of the call connection sequence of a micro ground station and a center station in the communication system by Embodiment 1 of this invention. It is a flowchart which shows the process example of the INVITE message of the SIP proxy part of the line multiplexing apparatus in the communication system by Embodiment 1 of this invention. It is a flowchart which shows the process example of the INVITE message of the SIP proxy part of the line multiplexing apparatus in the communication system by Embodiment 1 of this invention. It is a figure which shows the example of the INVITE message which the line multiplexing apparatus in the communication system by Embodiment 1 of this invention outputs. It is a figure which shows the example of the INVITE message which the line multiplexing apparatus in the communication system by Embodiment 1 of this invention outputs. It is a flowchart which shows the example of a process of 200OK message of the SIP proxy part of the line multiplexing apparatus in the communication system by Embodiment 1 of this invention. It is a figure which shows the example of the 200OK message which the line multiplexing apparatus in the communication system by Embodiment 1 of this invention outputs. It is a figure which shows the example of the 200OK message which the line multiplexing apparatus in the communication system by Embodiment 1 of this invention outputs. It is a figure which shows the call connection sequence between the microminiature ground stations in the communication system by Embodiment 1 of this invention.

Explanation of symbols

  1 center station, 2 ultra-small ground station, 3 ultra-small ground station, 4 artificial satellite, 5 public telephone network, 6 public telephone network, 10a, 10b, 10c IPCME domain, 11a, 11b, 11c line multiplexer, 12a, 12b, 12c IP phone, 13a, 13b IP-PBX, 14a, 14b, 14c Satellite gateway device, 15a, 15b, 15c Ethernet (registered trademark) cable, 21a SIP proxy unit, 21b SIP proxy unit, 22a Media proxy unit, 22b Media proxy part, 100 header part, 101 IP header, 102 UDP header, 103 RTP header, 104 payload part, 105 short packet, 106 header part, 107 payload part.

Claims (5)

The local IPCME (Circuit Multiplication Equipment for IP-based networks) domain indicating the multiplexing target area of the local station and the remote IPCME domain indicating the multiplexing target area of the other station are registered and received SIP (Session Initiation) Protocol), when the source of the message matches the local IPCME domain of the local station and the destination of the received SIP message matches the remote IPCME domain of the counterpart station, the received SIP message is transferred to the counterpart station, When the transmission destination of the SIP message received from the mobile station matches the local IPCME domain of the local station, the received SIP message is transferred to the transmission destination of the local station.
When a SIP message is received from the source of the local IPCME domain of the local station, an IPCME-Gateway header describing text data including the local IPCME domain of the local station is added to the SIP header of the received SIP message.
Call connection control by acquiring and registering the remote IPCME domain of the counterpart station described in the IPCME-Gateway header added to the SIP header when a SIP message is received from the remote IPCME domain sender of the counterpart station A SIP proxy unit that performs
Based on the call connection control by the SIP proxy unit, the IPCME packet received from the partner station is transmitted by multiplexing the RTP (Real-time Transport Protocol) packet of the voice data from the source of the own station and transferring the IPCME packet to the partner station. And a media proxy unit for separating RTP packets and transferring RTP packets to the transmission destination of the local station .   The communication apparatus according to claim 1, wherein the local IPCME domain and the remote IPCME domain registered by the SIP proxy unit are expressed in a domain or network.   The SIP proxy unit checks whether or not the status information of the communication device of the partner station is normal, and if there is an abnormality, returns a BUSY message response to the transmission source and rejects the call connection. The communication apparatus according to 1. In a communication system using a satellite line composed of a center station and a plurality of micro ground stations,
The communication device according to claim 1 is provided in the center station and the plurality of micro ground stations,
The microminiature ground station transmits a SIP message and voice data to the microminiature ground station of the counterpart station via the center station. 5. The communication system according to claim 4, wherein the communication device is installed in a satellite gateway device that is installed in the center station and the plurality of micro ground stations and performs communication with an artificial satellite.

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