KR20060034167A - Apparatus and method for supplying voip service using session initiation protocol in cable network - Google Patents

Abstract

According to an aspect of the present invention, there is provided a method for providing a VoIP service using a SIP protocol in a cable network, when receiving a VoIP call from the outside through a cable network, identifying a protocol used in the call from a type of a received message of the call Connecting the event handler and the corresponding protocol stack according to the identified protocol and setting up a call accordingly.If the VoIP call is sent out through the cable network, the user can use the terminal to control the media gateway control protocol and the SIP. Selecting one protocol and connecting an event handler to the corresponding protocol according to the selected protocol to perform call setup.

Description

Apparatus and Method for Supplying VoIP Service Using Session Initiation Protocol in Cable Network}             

1 is a block diagram of an HFC network for VoIP calls according to the present invention.

2 is a configuration diagram of an MGCP network.

3 is a structural diagram of a stack simultaneously supporting the MGCP and SIP protocols according to the present invention;

4 is a process flow diagram when a call based on the MGCP protocol according to the present invention is received.

5 is a process flow diagram when a call is received based on the SIP protocol according to the present invention.

6 is a flowchart illustrating a case of originating a call based on the SIP protocol according to the present invention.

* Description of the symbols for the main parts of the drawings *

11: EMTA 12: CMTS

13: Call Agent 14: MTA Prov. server

15: CM Prov. Server 16: SIP Server

30: MGCP / SIP status information storage unit 31: MGCP stack

32: SIP stack 33: MUX

34 event handler

The present invention relates to a packet cable standard for supporting a VoIP service in a cable network. More specifically, a VoIP service using a SIP protocol in a cable network supporting a SIP protocol without additional equipment in a packet cable compatible terminal supporting MGCP. The present invention relates to a providing apparatus and a method.

PacketCable, announced by CableLab, is one of the many approaches to providing Voice over Internet Protocol (VoIP) services over cable networks. It is a multimedia architecture proposed to provide better services on top of DOCSIS 1.1 cable modem architecture.

With the introduction of packet cables, cable service providers can not only provide new IP-based real-time multimedia services using cable networks, but also add security, QoS, and investment protection (IP) to the infrastructure of existing networks. ) Can be further enhanced. You can also implement VoIP to ensure interoperability between products from multiple vendors.

IP is widely used as a standard transport mechanism in today's data networks, enabling new Internet application services such as multimedia e-mail, live chat, streaming media with music and video, and video conferencing. Packet cable was also developed with the intention of providing cable service providers with a network architecture that can deliver these services faster and more economically.

PacketCable networks require DOCSIS 1.1-capable bidirectional HFC networks, IP backbones, networks such as PSTN, on-net voice and data communications, PSTN connectivity, CPE provisioning, and service billing. In order to implement Billing, etc., various components as described below are included.

Cable Modem Termination System (CMTS) is a device that provides connection to an IP backbone network while interworking with cable modems of cable subscribers in a hybrid fiber coax (HFC) network. Packet cable network CMTS requires dynamic QoS support.

The only requirements for customer premises equipment (CPE) on packet cables are cable modems and multimedia terminal adapters (MTAs) that are compatible with DOCSIS 1.1.

The MTA must support all the signaling and encapsulation features required for physical and network interfaces, codecs, VoIP transport and QoS signaling for voice equipment (eg, regular telephones). In general, MTAs and cable modems have a standalone MTA type that is separate from the cable modem and MTA (Embedded MTA), NID (Network Interface Device), and existing cable modem. There are three elements.

Packet-cable networks require a variety of different telephony components that use the Network-based Call Signaling (NCS) protocol. NCS is based on the Media Gateway Control Protocol (MGCP). Call control in the packet cable is controlled by a CMS (Call Management Server) called a soft switch. The CMS is a major component of the packet cable network, which coordinates setup and tear-down for all calls.

Media / Trunk gateways provide the communication path between the PSTN and packet-cable IP networks. Typically, T1 / E1 or T3 trunks are provided as physical interfaces to the PSTN.

The Media Gateway Controller (MGC) controls the media gateway, receives and coordinates call signaling information between the packet cable and the PSTN, and maintains and regulates the overall state of calls that require PSTN interconnection. . It also serves to convert voice bearer traffic from TDM voice to packetized voice (VoIP) and vice versa.

The signaling gateway plays a role of exchanging signaling with the PSNT network through the SS7 connection at the end of the packet cable network, and exchanging the signaling with the MGC / CMS based on the signaling gateway.

Media Server provides the functions or services except the CMS does. Announcement server notifies the caller of network notifications (for example, 'all lines are busy'). The Conference Server supports conferences such as n-way calling. Voice mail or Unified Messaging Server (UMS) can provide services that integrate voice and email.

The packet cable architecture requires a provisioning server and a mechanism for provisioning CPE equipment such as cable modems. You also need components to provision cable modems, such as Domain Name Server (DNS), Dynamic Host Configuration Protocol (DHCP), Trivial File Transfer Protocol (TFTP), and Time of Day (ToD) services.

Record-keeping server (RKS) is a hardware device that acts as a gateway to collect billing, providing an interface between billing systems for cable operators.

In the cable network as described above, MGCP is used as a signaling protocol for VoIP service. However, compared to MGCP, which has limited use, SIP is expanding, and thus the speed of spreading from messengers used in PCs to enterprise key phones and mobile communication services is increasing. Since SIP and MGCP are not compatible with each other, users need to replace or add a terminal in order to use another protocol at the same time.

In order to solve the above problems, the present invention proposes a method of simultaneously supporting the SIP protocol in EMTA, so that VoIP in the cable network using the SIP protocol can be used in the SIP without using additional equipment. It is an object of the present invention to provide an apparatus and method for providing a service.

VoIP service providing method using the SIP protocol in the cable network according to an aspect of the present invention, when receiving a VoIP call from the outside through a cable network, the step of identifying the protocol used in the call from the type of the received message of the call; Connecting an event handler for processing an event of a physical layer and a protocol stack identified in the step among various protocol stacks provided by the VoIP service, and setting up a call according to the corresponding protocol.

If the protocol identified in the protocol identification step is a media gateway control protocol, it checks whether the SIP protocol is in use and terminates an incoming call if there is an ongoing call using SIP, and if the SIP is not in use, the media gateway protocol. Call setup is performed by connecting the stack and event handlers.

If the identified protocol is SIP, check whether the media gateway control protocol is used, and if there is a call in progress using the media gateway control protocol, terminate the incoming call, and if the media gateway control protocol is not in use, The call setup is performed by connecting the media gateway protocol stack and the event handler.

In the method for providing a VoIP service using the SIP protocol in a cable network according to another aspect of the present invention, when a VoIP call is sent out through a cable network, a user selects one of a media gateway control protocol and a SIP using a terminal. And attaching an event handler to a corresponding protocol according to the selected protocol to perform call setup.

If the protocol used by the terminal is set based on a media gateway control protocol, the call setup is performed by switching to the SIP mode by selecting a preset button for switching the SIP mode on the terminal, and ending the call. Thereafter, the terminal returns to the media gateway control protocol mode again, and the SIP mode switching button can be set to a specific letter or number by an administrator.

In the cable network according to another aspect of the present invention, the VoIP service providing terminal apparatus using the SIP protocol monitors the state of the MGCP stack and the SIP stack included in the terminal, and monitors events related to call establishment with the media gateway control protocol. MGCP / SIP status information storage for determining which protocol stack of SIP to notify, and a MUX for connecting the generated call event to the protocol stack determined by the MGCP / SIP status information storage.

Hereinafter, a preferred embodiment according to the present invention will be described in detail with reference to the drawings.

1 shows an HFC network configuration for VoIP call according to the present invention.

Various servers (MTA Prov. Server 14, CM Prov. Server 15) are the equipment necessary for bringing the EMTA 11 online. The Embedded Multimedia Terminal Adapter (EMTA) is a terminal that combines a DOCSIS 1.1 (or later) compatible cable modem module with a MGCP / NCS-based VoIP call module.

The CMTS 12 is a DOCSIS 1.1 compatible headend device, and the call agent 13 is a call signaling device operating based on MGCP / NCS.

MGCP is a protocol that defines the communication between the media gateway and the call control element. The goal of MGCP is to keep it simple. It is a protocol implemented to reduce the role of the media gateway in the multi-service packet network converting audio signals and data packets, and to intelligently handle call processing and control in the call agent 13 or the media gateway controller.

The SIP server 16 needs to be added to the existing HFC network configuration to support SIP. The SIP server functions as a domain service, a user and a corresponding SIP URL or phone number management, and a location service that locates a terminal. Do it. Information about a user, a phone number, and a SIP URL can be shared by adding user information necessary for SIP to a DB used by an existing call agent 13. Phone numbers can be shared by MGCP and SIP.

2 shows the configuration of an MGCP network.

As shown in FIG. 2, in order to simplify the gateway structure and reduce the processing burden, a complex and intelligent call processing function is distributed to an external call processing device. As a result, the gateway provides a more advanced voice and video media service. To make it possible.

The MGCP performs its function with the sequential delivery of commands and requires acknowledgment of the delivered message. That is, the media gateway controller sends an MGCP command to the media gateway and receives an ACK for each command. Each message is an ASCII coded string. The command consists of a command called "verb" that must be performed by the command and additional parameters.

MGCP establishes a connection establishment message between two ends, CRCX (Create Connection), a connection disconnection message between two ends, DLCX (Delete Connection), a connection state change message from a gateway perspective, MDCX (Modify Connection), and a gateway according to each event occurrence. Message Request to send a notification to the call agent 13 Notification Request (RQNT), message NTFY (Notify) that the gateway sends information to the call agent 13 according to the occurrence of each event; The message that the gateway signals to the call agent 13 when it is restored. Restart In Progress (RSIP), the message that the call agent 13 collects the status information of the endpoints. The audit endpoint (AUEP), and the call agent 13 are connected. Message to collect information It consists of 8 commands of AUCX (Audit Connection).

Messages such as CRTX, DLCX, and MDCX related to the connection establish, release, or change the connection, and messages such as AUEP and AUCX related to the state modify the state information. Event-related messages include RQNT and NTFY. RSIP is a message caused by service failure or interruption.

In the present invention, the SIP protocol (Session Initiation Protocol) to be added in addition to the MGCP protocol is defined in the Internet Engineering Task Force (IETF) Request for Comments (RFC) 2543 document, and a session for multimedia communication such as video and audio ( Application layer control protocol for establishing, modifying, and terminating sessions or calls.

SIP is an application layer control protocol that creates, modifies, and releases end-to-end multimedia sessions, where the multimedia sessions can be multimedia conferences, Internet telephony calls, multimedia applications, and so on. . It is important to note that SIP is not a protocol that performs VoIP service at once by using a combination of protocols, but simply a protocol that deals with session establishment. In other words, SIP is simple and has the extensibility to be used with several other protocols.

SIP is a protocol that exists above the UDP / TCP / IP layer. It is a client / server protocol that can send and receive SIP request and SIP response messages in a request / response manner. It supports both unicast and multicast sessions to get started.

SIP Request messages can be received in RFC2543 by INVITE (Invite to Join Session), ACK (Acknowledge Request), BYE (End Call), REGISTER (User Agent registers in the redirect server's database), and CANCEL (Cancel pending requests). ), 6 basic functions of OPTIONS are provided, and SIP Response message is 1xx (information response), 2xx (successful response), 3xx (redirect response), 4xx (client error, request failure), 5xx as status code. (Server failure), 6xx (global failure).

EMTA 11 according to the present invention is a terminal operating based on MGCP / NCS, but also includes a module for SIP protocol processing. There is no need for a separate hardware change of the EMTA 11 in accordance with SIP support. The EMTA 11 may support only one channel or may support multiple channels. Since the operation of the EMTA 11 is independent of the number of channels, the present invention will be described by taking the EMTA 11 mainly supporting one channel as an example.

In the present invention, the MGCP / SIP Status Info storage unit 30 is used as shown in Fig. 3 for use in parallel with the SIP protocol.

3 shows a structure of a protocol stack for simultaneously supporting the MGCP and SIP protocols according to the present invention.

The MGCP stack 31 and the SIP stack 32 are connected to the application, and the lower end of the event handler 34 is connected to the physical layer.

The event handler 34 is a part that handles event generation that may occur in mechanical parts such as hook detect, ringing, and tone generation of the handset.

The MUX 33 serves to connect the selected one of the MGCP stack 31 and the SIP stack 32 to the event handler 34.

The role of the MGCP / SIP state information storage unit 30 is to monitor which state each protocol (MGCP, SIP) is in and determine which protocol stack should be notified of all events related to phone settings. . This is done by notifying the currently active protocol stack of events (hook on / off, DTMF, etc.) generated by activating the MUX of FIG.

The present invention enables basic call setup through the protocol stack shown in Figure 3.

First, if you are dialing, you need to decide which protocol, MGCP or SIP, will use the VoIP service. For this purpose, a specific number or letter is assigned to MGCP and SIP, and the user can select MGCP or SIP by pressing the number or letter corresponding to each service.

In case of a call, the EMTA 11 located on the user side will be the receiving EMTA 11. Therefore, the first message received by the EMTA 11 from another network will be "CRCX" (Create Connection), which is a connection establishment message between two ends in case of MGCP, and call establishment from either side of the network in case of SIP. It will be an "INVITE" message requesting a. If so, the EMTA 11 analyzes the first message received from the network to determine which protocol is the VoIP service provided by the protocol, and then sets or changes the mode to be processed by the EMTA 11.

4 shows a processing flow when a call is received based on the MGCP protocol according to the present invention.

In the idle state (S41), when a call is received (S42), EMTA (11) once to determine what protocol the incoming call is using (S43). This process is performed by identifying both the MGCP protocol and the SIP protocol, and then comparing which protocol the received message corresponds to which protocol.

As described above, the MGCP message includes CRTX, DLCX, MDCX related to connection, AUEP, AUCX related to status information, RQNT related to event, NTFY, and RSIP message related to service failure or interruption. On the other hand, SIP has six kinds of request messages: INVITE, ACK, BYE, REGISTER, CANCEL, OPTIONS, and SIP response messages include 1xx (information response), 2xx (successful response), 3xx (redirect response), 4xx (client error, request failure), 5xx (server failure), 6xx (global failure) are provided.

Since FIG. 4 is a case where a call is received, a CRTX (Create Connction) message for the MGCP protocol and an INVITE (invitation) message will be received for the SIP protocol.

After identifying which protocol based call is received from the incoming message, you should check the status of any protocol other than that protocol. In FIG. 4, since it is assumed that a call using the MGCP protocol is received, the state of the SIP protocol stack should be checked (S44). Check the status of the SIP protocol to the MGCP / SIP status information storage unit 30.

If the SIP stack is in use, call processing cannot be performed using MGCP, so it is checked whether SIP is in use (S45). If the SIP stack is not in use, the MUX 33 may be switched (S46) so that the MGCP stack and the event handler 34 may be connected (S46), and the general MGCP call processing may be performed (S48).

If the SIP is in use, two or more calls cannot be simultaneously carried out through one EMTA, so that the call received based on the MGCP is terminated (S47). At this time, the EMTA 11 transmits a 5XX error message meaning 'no resource available' to the call agent 13.

5 shows a processing flow when a call is received based on the SIP protocol according to the present invention.

FIG. 5 is similar to the case of FIG. 4, but assumes that a call based on the SIP protocol is received by the EMTA 11 rather than the MGCP protocol.

Basically, in the idle state (SLE), the call is received (S52) and the same until the step (S53) of determining the protocol of the received call. However, in the case of FIG. 5, since the SIP protocol is used, it will be confirmed that the SIP protocol is a result of the determination of the received protocol. Therefore, the EMTA 11 will have to check the state of the MGCP stack, not SIP (S54).

As a result of checking the status of the MGCP stack, when the MGCP is not in use, the MUX is switched to the SIP stack (S56), and general SIP call processing may be performed (S58). If the MGCP stack is in use, it terminates the call that was received based on the SIP, and transmits an error message of 5XX for server failure or 6XX for global failure to the caller (S57).

6 shows a processing flow when a call is made based on the SIP protocol according to the present invention.

Add a specific option in the configuration file for bringing the EMTA 11 online so that the number to switch to the SIP mode can be set. In FIG. 6, if '#' is selected before the incoming telephone number is selected, this means that the switch means the SIP mode. Basically, EMTA (11) was considered to be processed based on MGCP, and configured to press an additional selection button if desired to switch to SIP mode.

However, this is an embodiment, and if the SIP mode is more commonly advertised, the SIP mode may be set as the default and MGCP may be optional, and the SIP toggle button may be assigned to a specific letter or number rather than '#'. It is also possible to set it. Therefore, even if the form is different from the embodiment shown in Figure 6, if the basic selection key is the same, it will be considered an invention belonging to the scope of the present invention.

In order to send a call using the SIP protocol, once the MGCP protocol is set to idle (IDLE) state by default (S601), the user starts from the step of picking up the handset for the call (S602). At the time of picking up the handset, the protocol for the call is set to MGCP.

The user who picks up the handset selects a button in which the function for switching to the SIP mode is set (S603). In the example of Figure 6 it will be necessary to press the '#' button. The moment the user selects the '#' button, the EMTA 11 switches the MUX to the SIP stack (S604). The 'SIP in use' field is set to TRUE (S605).

An INVITE message of the SIP protocol requesting a call connection to the called party is transmitted to the receiver (S607), and when the called party responds, a call is made to connect the calling party and the calling party. If the call lasted for a certain time (S608), the 'SIP in use' field is changed to FALSE (S609).

The MUX having moved the connection setting to the SIP stack is switched back to the MGCP stack (S610). The EMTA 11 is placed in the IDLE state again, and the default protocol is set to MGCP (S611).

On the other hand, although not shown, it is also possible to consider the case of sending a call based on the MGCP, in this case, considering that the existing EMTA (11) is basically packet cable compatible, the characteristic element according to the present invention As there is no additional part, it can be understood to operate by the existing packet cable NCS standard.

The present invention enables both a cable operator and a user who wants to receive VoIP service using a cable network to simultaneously support SIP and MGCP at a minimum cost, thereby helping to expand the VoIP service as well as economic effects.

Claims (12)

In the VoIP service providing method in a cable network, If a VoIP call is received from the outside through the cable network, Identifying the protocol used in the call from the type of received message of the call; Wow Using a SIP protocol in a cable network comprising the step of connecting the event handler for processing events of the physical layer and the protocol stack identified in the step of the various protocol stacks provided by the VoIP service, and establishing a call according to the corresponding protocol How to provide VoIP service. The method of claim 1, If the protocol identified in the protocol identification step is a media gateway control protocol, Check whether the SIP protocol is used, If there is an ongoing call using SIP, the incoming call is terminated. If the SIP is not in use, the VoIP service providing method using the SIP protocol in the cable network, characterized in that the call connection is performed by connecting the media gateway protocol stack and the event handler. The method of claim 2, The media gateway protocol message of the received call is a CRCX, VoIP service providing method using a SIP protocol in a cable network. The method of claim 1, If the protocol identified in the protocol identification step is SIP, Check whether the media gateway control protocol is used, If there is an ongoing call using the media gateway control protocol, the incoming call is terminated. If the media gateway control protocol is not in use, the VoIP service providing method using the SIP protocol in a cable network, characterized in that the call setup is performed by connecting the media gateway protocol stack and the event handler. The method of claim 4, wherein The SIP protocol message of the received call is INVITE, VoIP service providing method using the SIP protocol in a cable network. In the VoIP service providing method in a cable network, When outgoing VoIP calls over a cable network, A user selecting one of a media gateway control protocol and a SIP using a terminal; Wow And connecting the event handler to the protocol according to the selected protocol to perform call setup. The method of claim 6, If the protocol used by the terminal is set based on the media gateway control protocol, In the cable network, the terminal returns to the media gateway control protocol mode by performing a call setup by switching to the SIP mode by selecting a preset button for switching the SIP mode on the terminal. Method of providing VoIP service using SIP protocol. The method of claim 7, wherein The SIP mode switch button is a VoIP service providing method using the SIP protocol in the cable network, characterized in that the administrator can be set to a specific letter or number. An apparatus having both an MGCP stack and a SIP stack and providing VoIP service, An MGCP / SIP state information storage unit configured to monitor the state of the MGCP stack and the SIP stack included in the terminal, and determine which protocol stack of the media gateway control protocol and SIP is notified of an event related to call establishment; Wow A VoIP service providing terminal apparatus using a SIP protocol in a cable network including a MUX for connecting the generated call event to the protocol stack determined by the MGCP / SIP state information storage unit. The method of claim 9, If a VoIP call is received from the outside through the cable network, From the type of received message of the call to determine whether the protocol used in the call is a media gateway control protocol and SIP, The VoIP service providing terminal apparatus using the SIP protocol in the cable network, characterized in that for connecting the event handler and the corresponding protocol stack according to the identified protocol and set the call accordingly. The method of claim 10, When outgoing VoIP calls over a cable network, In the cable network, a user selects one of a media gateway control protocol and a SIP using a terminal, and sets up a call by connecting an event handler to a corresponding protocol according to the selected protocol. Providing terminal device. The method of claim 11, If the protocol used by the terminal is set based on the media gateway control protocol, Switching to SIP mode is performed by selecting a button set for SIP mode switching on the terminal, and after the call is terminated, the terminal returns to the MGCP mode. VoIP service providing terminal device.

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