JP2006340260A - Call control method of internet telephone - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To enable an Internet telephone to be used by a server provided by an Internet telephone carrier even when a standard telephone is used under an environment of an existing fire wall without correcting content of an SIP telephone. <P>SOLUTION: In a control system 100 of an Internet telephone, an SSC server 6-3 which controls an SIP message on a caller side and a voice signal control protocol (RTP) are managed before an SIP proxy server 6-1 on a server side which becomes an existing officially recognized IP network, an MRM server 6-4 which generates an RTP IP port is arranged, an SDP message is converted into an address port in which a caller and an MRW server 6-5 are bound by the management of the MRM server 6-4, an RTP packet is relayed from the MRW server to the SIP proxy server 6-1, and a call with an Internet telephone 11-1 within a private IP network is enabled through an IP shared device 1 (fire wall) from the officially recognized IP network side. <P>COPYRIGHT: (C)2007,JPO&INPIT

Description

  The present invention is a SIP (Session Initiation Protocol) proxy that forms a local network on the main side so that a private IP network and an authorized IP network are connected via an IP sharing machine and a firewall, and functions such as call routing are performed in the authorized IP network. The present invention relates to a call control method for an Internet telephone using a server.

  More specifically, in order to solve the shortage of IP for a call in a VoIP environment using SIP, there is a single IP address in the public IP network so that the IP sharing machine to be used can be applied in the VoIP environment. A method for passing through an IP sharing machine or a firewall through a local network on the main side of a standard Internet telephone, which enables a call with a general telephone connected to a plurality of other IP telephones (Internet telephones) and a public telephone network (PSIN). The present invention relates to a call control method for an Internet telephone having features.

  In general, the use of Internet telephones is increasing as the environment using the Internet is established. FIG. 1 shows an operating environment using an existing Internet telephone (VoIP). As shown in this figure, the operating environment is roughly divided into a local network (LAN), a wide area information communication network (WAN), and a main local network (LAN). The local network (LAN) is divided into a private IP environment and a certified IP environment, and each environment is separated by the VoIP IP sharer 1.

  The private IP environment is connected by a private IP between the IP sharing device 1 and the VoIP telephone 2. The certified IP environment is taken over by the IP sharing machine 1 for VoIP that makes a boundary with the private IP environment, the router 3 that connects the call including the IP address by the VoIP sharing machine 1, and the Internet wide area information communication network (WAN). It comprises a digital service unit 4 which converts and sends it to a setting bipolar signal.

  The main local network (LAN) is a certified IP environment, and is connected to a digital service unit 4-1 that converts a bipolar signal from a wide area information communication network (WAN) into a unipolar signal and a certified IP environment. The router 3-1 for setting the route, the VoIP exchange 6 and the VoIP gateway 7 connected by the hub 5 so as to recognize the IP based on the route via the router 3-1, and the call via the VoIP gateway 7 Are connected to a public telephone network (PSTN) that connects to the general telephone 8 and a VoIP telephone 2-1 connected to the hub 5.

  In such an operating environment, the VoIP call method is such that the two terminals (VoIP phones) 2 and 2-1 include their own IP addresses in the call signal to the other party in order to make a VoIP call. send. The signal receiver responds to the IP address of the signal source included in the call signal. In this case, when a private IP is used, the IP address for the VoIP telephone in the private IP environment is an IP address (Unknown IP) that is unknown to the VoIP telephone in the certified IP environment. Impossible.

  In order to solve this problem, the IP sharing machine 1 for VoIP is separately manufactured and used in place of the existing equipment. The VoIP IP sharing machine 1 converts a private IP address in the call signal generated by the VoIP telephone 2 in the private IP environment into an authorized IP address of the IP sharing machine and retransmits the call signal. Is the main purpose. The call signal thus converted can be recognized by the VoIP telephone 2-1 in the certified IP environment.

  However, when using an IP sharing machine for VoIP, an additional process for the unique function of the IP sharing machine is required, causing a delay in the network and reducing the VoIP call quality. The reason for this is that a general IP sharing machine only performs a function of converting a private IP address into an authorized IP address, whereas a VoIP IP sharing machine does not handle all IP packets that pass through the IP sharing machine. This is because filtering and updating of data for the VoIP signal are performed.

  Further, although there is a demand for a layer change for the structure of the IP sharing device, the network layer in a general sharing device environment is as shown in FIG. As shown in this figure, the VoIP telephone 2-2, the IP sharing machine 1-1, and the router 3-2 connected to the Internet network have a 5, 3, and 3 layer structure.

  FIG. 3 is a diagram showing a network layer in the VoIP IP sharing machine 1 environment. As shown in this figure, the VoIP telephone 1, the IP sharing device 2, and the router 3 connected to the Internet network have a 5, 5 and 3 layer structure.

  Therefore, the standard sharing machine of the general concept proceeds in layer 3 (Layer 3) as shown in FIG. 2 by the IP sharing machine 1-1, and the IP sharing machine 1 for VoIP uses layer 5 as shown in FIG. Proceed with (Layer5). For this reason, the IP sharing machine 1 for VoIP requires a separate additional time (Overhead), which acts as a serious drawback from the viewpoint of network management.

  In addition, even if the standard shared device is replaced with a VoIP shared device, additional operation costs still occur, and compatibility between the VoIP shared device and the standard shared device may also be problematic. In addition, since it is necessary to install a shared device for each VoIP protocol, problems arise in maintenance and operation.

  On the other hand, SIP is a protocol used in a server (VoIP exchange), which is a protocol for transmitting media (including telephones) on the Internet protocol defined by the IETF (Internet Engineering Task Force) organization. It is. This is a protocol (Application Layer Protocol of OSI 7 Layer) that can set, change, and terminate a session such as an Internet telephone call (call), and is an ITU-T used as an existing Internet telephone service protocol. Compared to the standard defined by H.323 (H.323), the call setup time is short, and it has a technically superior architecture that can support various additional services with flexibility. It is a communication protocol that can be used as a protocol for services and additional services such as a voice mail box.

  Here, in an environment where a private IP network is installed, a technology has been developed that enables VoIP calls without changing the setting of existing equipment in a VoIP environment using SIP. FIG. 4 shows a usage environment of a VoIP telephone using a SIP server.

  As shown in this figure, the usage environment of a VoIP telephone using SIP is roughly divided into a local network (LAN), a wide area information communication network (WAN), and a main-side local network (LAN). The local network (LAN) is divided into a private IP environment and a certified IP environment, and each environment is separated by the standard IP sharing machine 1-1. The private IP environment is connected by a private IP between the IP sharing machine 1-1 and the SIP telephone 11. The certified IP environment includes an IP sharing machine 1-1 that forms a boundary with a private IP environment, a router 3 that recognizes an IP address via the IP sharing machine 1-1, and connects to a path, and an Internet wide area information communication network (WAN). The digital service unit 4 converts it into a set bipolar signal and sends it to take over. As shown in FIG. 2, the IP sharer 1-1 has an IP sharer 1 composed of layer 3 so that data (TCP / IP) is transmitted separately in a private IP network (NAT). -1.

  The main local network (LAN) is a certified IP environment, and is connected to a digital service unit 4-1 that converts a bipolar signal from a wide area information communication network (WAN) into a unipolar signal and a certified IP environment. Connected by the hub 5 to the server 6, 6-1, 6-2 connected by the hub 5 so as to recognize the IP based on the router 3-1 that sets the path and the path through the router 3-1. And a SIP gateway 7 connected to the exchange PBX for switching between the general telephone 8 and the public telephone network PSTN.

  However, the VoIP call method using SIP in such an operating environment is a method that enables a call in a VoIP environment, but cannot be applied to all standard telephones used in a firewall or a private IP network. There was a problem.

  An object of the present invention is to make an Internet telephone call that enables an Internet telephone to be used by a server provided by an Internet telephone carrier even if a standard telephone is used without modifying the contents of the SIP telephone in an existing firewall environment. It is to propose a control method.

  In order to solve the above-described problems, the Internet telephone call control method according to the present invention includes an SSC server for controlling a SIP message on the caller side and a voice before the SIP proxy server on the server side constituting the existing certified IP network. The MRM server that manages the signal control protocol (RTP) and generates the RTP IP port, and the management of the MRM server converts the SDP message to the bound address port of the caller and the MRW server, and the MRW server converts the RTP packet. A system configuration that relays to a SIP proxy server is adopted so that a call can be made from an authorized IP network side to an Internet telephone in a private IP network through an IP sharing device and a firewall.

That is, the present invention connects a private IP network and an authorized IP network through an IP sharing device and a firewall, and performs SIP (Session Initiation Protocol) on the main local network so as to perform functions such as call routing in the authorized IP network. In a telephone telephone call control method using a proxy server,
An SSC server (SIP Signaling Control Server) is added in front of the SIP proxy server,
A user of an Internet telephone (SIP telephone) that is a cola designates its own IP and port, and in the form of binding, registers the IP and port of the Internet telephone as a user in the SSC server,
The SSC server stores contact head information, route (Record-Route) information, source (user) address, and port number, encodes this with a message, and transmits the encoded message to the SIP proxy server. Wait for a response
Upon receiving the response, the SSC server decodes the response message, and calls the Internet telephone user through the IP sharer and the firewall based on socket information of the registered port (IP PORT). It is characterized by being able to do it.

In addition, the present invention uses a SIP proxy server that forms a local network on the main side so as to connect a private IP network and a certified IP network via an IP sharing device and a firewall, and perform functions such as call routing in the certified IP network. In the Internet telephone control method,
The signal transmitted to the SIP proxy server is taken into an NFT server (NAT & Firewall Traversal Server),
The NFT server receives a standard SIP message from an Internet telephone in the private IP network, encodes / decodes the standard SIP message, and then passes the decoded message through the IP sharer and the firewall. It is characterized by performing processing for transmission to an Internet telephone.

In the Internet telephone call control method of the present invention, the NFT server includes an SSC server (SIP Signaling Control Server), an MRM server (Media Relay Maser Server), and an MRW server (Media Relay Worker Server). The server requests the MRM server to generate an RTP IP port (Real-time Transport Protocol IP Port),
The MRM server requests port binding to the MRW server, and the MRM server receives a response from the MRW server and responds to the SSC server. The SSC server includes a caller-side Internet telephone (SIP telephone) and the MRW server. The server bind address is converted (decoded / encoded) into SDP (Session Description Protocol), and then an invite message is sent to the SIP proxy server.

  In the Internet telephone call control method of the present invention, when ringing from the SIP proxy server to the SSC server is completed, the SSC server requests the MRM server for the IP and port of the MRW, and the MRM server The MRW server IP and port are transmitted to the SSC server, and the SSC server converts (decodes / encodes) a bound address of the Internet phone on the callee side and the MRW server into SDP, and It is characterized by letting pass.

  Furthermore, in the call control method for an Internet telephone according to the present invention, when the caller-side Internet telephone receives the address, it sends media data to a bound IP port of the MRW server, and the MRW server transmits the media data to the callee. The callee-side Internet telephone sends the media data to the caller-side Internet telephone via the bound IP port of the MRW server.

  According to the call control method of the present invention, call processing is performed in the NFT server including the SSC server, the MRM server, and the MRW server, and the media data relay during the call is shared by the MRW server, so that the call traffic is reduced. It is possible to improve the call quality by distributing.

  Further, by supporting a standard SIP telephone in an Internet telephone, a user can purchase a telephone and use it for general purposes regardless of the manufacturer.

  Further, it is economical for the Internet telephone service provider (ITS) because the terminal can be standardized and the development cost of an additional program for passing through the firewall can be reduced.

  Embodiments of the present invention will be described below with reference to the drawings.

  FIG. 5 is a schematic configuration diagram showing an example of a telephone telephone call control system to which the present invention is applied. The call control system 100 includes a local network (LAN), a wide area information communication network (WAN), and a main local network (LAN). The local network (LAN) is divided into a private IP environment and a certified IP environment, and the environment is separated by a standard IP sharing machine 1-1. The private IP environment is connected by a private IP between the IP sharing machine 1-1 and the SIP telephone 11. The certified IP environment includes an IP sharing machine 1-1 that forms a boundary with a private IP environment, a router 3 that recognizes an IP address via the IP sharing machine 1-1, and connects to a path, and an Internet wide area information communication network (WAN). The digital service unit 4 converts the unipolar signal into a set bipolar signal and transmits it so as to take over. As the IP sharing machine 1-1, as shown in FIG.

  The main local network (LAN) forms a certified IP environment, and a digital service unit 4-1 that converts a bipolar signal from a wide area information communication network (WAN) into a unipolar signal, and a connection path in the certified IP environment. Are connected by the hub 5 and the servers 6, 6-1 and 6-2 connected by the hub 5 so as to recognize the IP based on the route through the router 3-1. SIP telephone 11-1 and SIP gateway 7 connected to the exchange PBX for switching the general telephone 8 and the public telephone network PSTN. Further, the main-side local network has an NFTS server connected between the SIP proxy server 6-1 and the hub 5.

  FIG. 6 is a schematic configuration diagram showing the main part of the call control system 100 of this example. As shown in this figure, the NFT server 6-0 (NAT & Firewall Traversal Server) is divided into an SSCS server 6-3, an MRMS server 6-4, and an MRWS server 6-5. An SSC server (SIP Signaling Control Server) 6-3 is a server that controls SIP messages, an MRM server (Media Relay Maser Server) is a server that manages RTP channel allocation / cancellation, and an MRW server (Media Relay Worker). Server) is a server that opens an RTP channel and relays RTP packets.

  The SSC server 6-3 requests the MRM server 6-4 to generate the RTP IP PORT. The MRM server 6-4 requests port binding from the MRW server 6-5. Further, the MRM server 6-4 receives the response of the MRW server 6-5 and responds to the SSC server 6-3. The SSC server 6-3 converts (decodes / encodes) the bound addresses of the cola-side Internet telephone (SIP telephone) 11 and the MRW server 6-5 into SDP (Session Description Protocol), and then converts the invite message to the SIP. Send to proxy server 6-1.

(User registration and firewall passage process)
FIG. 7 is a flowchart illustrating user registration by the call control system 100 of this example, and FIG. 8 is a flowchart illustrating user registration and firewall passage.

  To explain in accordance with these flowcharts, an SSC server 6-3 is added in front of the SIP proxy server 6-1 forming the main local network, and the user UA1 of the Internet telephone (SIP telephone) 11 serving as a cola The IP and port of the Internet telephone 11 are registered as a user in the SSC server 6-3 in the form of binding by specifying the port (R1). The SSC server 6-3 stores contact head information, route information, source (user) address, and port number, and encodes them with a message (R2). Then, the SSC server 6-3 transmits the encoded message to the SIP proxy server 6-1 (R3). When the SSC server 6-3 receives the response 200 OK from the SIP proxy server 6-1 (R4), it decodes the message, passes through the firewall based on the socket information of the registered port (IP PORT), and transmits the Internet. The user of the telephone 11 can talk (R5, R6).

  Next, the process of user registration and firewall passage will be described in more detail.

  The SIP telephone 11 of the user UA1 located inside the private IP or firewall sets the server address of the SSC server 6-3 (for example, 61.74.195.156) in the proxy server 6-1 and sends all SIP messages to the SSC server 6 -3. Then, the SSC server 6-3 confirms the message received from the user UA1, and transmits it to the corresponding proxy server 6-1.

  However, transmitting the message received from the user UA1 to the proxy server 6-1 may be a private IP environment or an authorized IP environment, but to send the message received from the proxy server 6-1 to the user UA1. First, when the user UA1 performs registration, the message of the user UA1 must be encoded (message conversion) (R1, R2).

  Here, it is confirmed whether or not the user ID at the time of encoding is an ID registered in the SSC server 6-3 in the message information (header information (Contact-Header)). If it is a registered ID, the user information is updated. If the user is not registered, the user is newly added to a local database (not shown, for example, a memory database). In this case, information to be registered when one user UA1 is added to the local database includes a header (From Header) which is a key of SIP encoded information and a source authorized IP address of the receiving socket (for example, 192.168.1.188 shown in FIGS. 7 and 8), source port number (UDP 5060 PORT), registration time, expire header, and the like.

  Therefore, based on the registered user UA1 information, the SSC server 6-3 transmits the encoded message to the proxy server 6-1 (R3), and the proxy server 6-1 transmits the message (for example, INVITE, (Including BYE, CANCEL, etc.) are sent to the SSCS server (R4).

  Then, the SSC server 6-3 decodes the message (R5), searches for the registered user UA1, and uses UDP to the mapped IP (for example, 192.168.1.188) and port number (for example, 5060). A SIP response message is transmitted.

  As described above, the SSC server 6-3 solves the problem of transmission and reception of SIP messages between the private IP and the proxy server 6-1 in the firewall environment.

(RTP session establishment and maintenance process for calls)
When the user is registered as described above, a process of maintaining a session is performed. That is, a UDP port opened in a private IP network or a firewall is closed if it is not used for a certain period of time. Therefore, in order to maintain communication with the SSC server 6-3, a periodic message (for example, UDP Keep- alive message) must be sent. An example of this is sending data (“CRLF) to the registered port (5060) that was transmitting at regular time intervals (eg, 10 second intervals).

  FIG. 9 is a flowchart showing a process of establishing and maintaining an RTP session for a call in this example, and FIG. 10 is a detailed flowchart thereof. It is assumed that the user UA1 using the SIP telephone 11 has requested a call. This request passes through the IP sharing machine (firewall) 1-1, passes through the router 3 and the digital service unit 4 in the authorized IP environment, and then passes through the Internet network that forms a wide area information communication network (WAN). -1. The digital service unit 4-1 converts the received signal into a unipolar signal, and then transmits it to the SSC server 6-3, which is an NFTS server, via the router 3-1, and the Internet telephone 11-1 of the Corey user UA 2 is transmitted. Call (S1).

  The SSC server 6-3 requests the MRM server 6-4 to generate an RTP IP PORT (IP PORT REQ) (S2). The MRM server 6-4 requests port binding to the MRW server 6-5 (PORT Bind REQ) (S3). Upon receiving the response (PORT Bind RES) from the MRW server 6-5, the MRM server 6-4 responds to the SSC server 6-3 (IP PORT RES) (S4, S5). The SSC server 6 converts (decodes / encodes) the bound addresses of the cola-side Internet telephone (SIP telephone) 11 and the MRW server 6-5 into SDP, and then sends an invite message to the SIP proxy server 6-1. (S7).

  Subsequent to this process (S7), when the ringing (180 Ringing) from the proxy server 6-1 to the SSC server 6-3 is completed (200 OK) (S8, S9), the SSC server 6-3 becomes the MRM server 6-4. Requesting the IP and port of the MRW server 6-5 (IP PORT REQ) (S10). The MRM server 6-4 transmits the IP and port of the MRW server 6-5 to the SSC server 6-3 (IP PORT RES) (S11). The SSC server 6-3 converts (decodes / encodes) the bound addresses of the Internet phone 11-1 on the Corey side and the MRW server 6-5 into SDP and passes through the firewall (IP sharing machine 1-1). (S12).

  Subsequent to the step (S12), the cola-side Internet telephone 11 receives this and sends the media data to the bound IP port of the MRW server 6-5 (S13). The MRW server 6-5 sends this to the Corey side Internet telephone 11-1 (S14). The callee-side internet telephone 11-1 sends the media data to the caller-side internet telephone 11 via the bound IP port of the MRW server 6-5 (S15).

  More specifically, when the SSC server 6-3 receives a message (INVITE) from the user UA1 located inside the private IP or firewall (S1), the value of the Call-ID header of the SIP message is used as key data. And registered in a call connection local database (not shown because it can be embodied in various forms as a memory).

  The registered message requests the MRM server 6-4 for the address and UDP port number of the MRW server 6-5 in charge of the RTP session (S2), and at the same time, the MRW server 6-5 establishes a UDP socket with the corresponding port number. After binding, it enters a waiting state (S3).

  On the other hand, the MRM server 6-4 has the address and port number of the MRW server 6-5 that can be used, and the port number is transferred from the MRM server 6-4 in the header ('m =') of the SDP message. Switch to the received port number (S4), replace the IP address of the SDP header ('c =') with the address received from the MRM server 6-4 (S5), and further encode the SIP and SDP messages The data is transmitted to the proxy server 6-1 (S6, S7).

  At this time, when the callee-side user UA2 receives the message (INVITE), the proxy server 6-1 sends a message (ACK) including the SDP body to the proxy server 6-1. The message is sent from the proxy server 6-1 to the SSCS server 6-1 (S8, S9). In addition, the SSC server 6-3 encodes the received message (ACK), reads the value (Call-ID header value), and changes the message by changing the SDP in the previous message (INVITE) (S10 , S11, S12), a message (ACK) is transmitted to the user UA1.

  As a result, RTP audio and video packets continue to be sent between the user UA1 who made the call and the user UA2 of the callee terminal that received the call, with the IP address designated by the SSC server 6-3 as the port number. A state is formed. Here, the designated IP address is the address of the MRW server 6-5, and the MRW server 6-5 receives the RTP packets entered from the user UA1 and the callee-side user UA2, but has received the first packet. At this time, the RTP packet exchange work is performed based on the socket information, so that the SIP terminal (SIP telephone) inside the private IP or the firewall can also make a voice / video call.

(Call termination process)
The call termination process is as follows. Upon reception of the message (BYE) (CANCEL), the header (Call-ID header) registered before is analyzed, and a command to release the UDP port bound to the MRW server 6-5 is sent. Also, the record (record registered in the local database generated with Call-ID) is removed. After this, it waits for the next call.

It is a block diagram showing a call connection network of a general VoIP telephone. It is a block diagram which shows the network layer in a general IP sharing machine environment. It is a block diagram which shows the network layer in IP sharing machine environment for VoIP. It is a system block diagram which shows the conventional use environment. It is a schematic block diagram which shows the example of a system to which this invention is applied. It is a schematic block diagram which shows the principal part of the system of FIG. It is a flowchart which shows the user registration process in the system of FIG. 6 is a flowchart showing a user registration and firewall passage process in the system of FIG. 6 is a flowchart showing a process of establishing and maintaining an RTP session for a call in the system of FIG. 10 is a detailed flowchart of the process of FIG.

Explanation of symbols

1-1 Standard IP sharing machine (firewall)
3, 3-1 Router 4, 4-1 Digital service unit 5 Hub 6-0 NFT server 6-1 SIP proxy server 6-3 SSC server 6-4 MRM server 6-5 MRW server 11 Internet telephone 11 on the caller side 1 Cory Internet Phone 100 Call Control System UA1 Cola User UA2 Cory User

Claims (5)

An Internet telephone that uses a SIP (Session Initiation Protocol) proxy server on the main local network so that a private IP network and an authorized IP network are connected via an IP sharing machine and a firewall, and functions such as call routing are performed in the authorized IP network. In the call control method of
An SSC server (SIP Signaling Control Server) is added in front of the SIP proxy server,
A user of an Internet telephone (SIP telephone) that is a cola designates its own IP and port, and in the form of binding, registers the IP and port of the Internet telephone as a user in the SSC server,
The SSC server stores contact head information, route (Record-Route) information, source (user) address, and port number, encodes this with a message, and transmits the encoded message to the SIP proxy server. Wait for a response
Upon receiving the response, the SSC server decodes the response message, and calls the Internet telephone user through the IP sharer and the firewall based on socket information of the registered port (IP PORT). A telephone call control method for an Internet telephone, characterized by being made possible. In a method for controlling an Internet telephone using a SIP proxy server that forms a local network on the main side so that a private IP network and a certified IP network are connected via an IP sharing machine and a firewall, and functions such as call routing are performed in the certified IP network. ,
The signal transmitted to the SIP proxy server is taken into an NFT server (NAT & Firewall Traversal Server),
The NFT server receives a standard SIP message from an Internet telephone in the private IP network, encodes / decodes the standard SIP message, and then passes the decoded message through the IP sharer and the firewall. A call control method for an Internet telephone, characterized by performing processing for transmission to an Internet telephone. In claim 2,
The NFT server includes an SSC server (SIP Signaling Control Server), an MRM server (Media Relay Maser Server), and an MRW server (Media Relay Worker Server),
The SSC server requests the MRM server to generate an RTP IP port (Real-time Transport Protocol IP Port),
The MRM server requests port binding from the MRW server,
The MRM server receives the response of the MRW server and responds to the SSC server;
The SSC server converts (decodes / encodes) the bound addresses of the Cola-side Internet telephone (SIP telephone) and the MRW server into SDP (Session Description Protocol), and then sends an invite message to the SIP proxy server. A telephone call control method for an Internet telephone. In claim 3,
When ringing from the SIP proxy server to the SSC server is complete, the SSC server requests the MRM server for the IP and port of the MRW,
The MRM server communicates the IP and port of the MRW server to the SSC server;
The Internet telephone call control method, wherein the SSC server converts (decodes / encodes) a bound address of a Cory-side Internet telephone and the MRW server into an SDP and passes the firewall. In claim 4,
Upon receipt of the address, the caller-side Internet telephone sends media data to the bound IP port of the MRW server,
The MRW server sends this to the Corey side internet telephone,
A call control method for an Internet telephone, wherein the Cory-side Internet telephone sends the media data to the Cola-side Internet telephone via the bound IP port of the MRW server.

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