Internet protocol transport of pstn-to-pstn telephony services

Description: TECHNICAL FIELD [Detailed description of the invention]

【0001】

[Technical field to which the invention belongs]

The present invention relates broadly to the field of telecommunications, and more specifically to transport public switched telephone network (PSTN) methods and systems in which signals are terminated through an Internet Protocol (IP) network. [0002]

[Problems to be Solved by Conventional Techniques and Inventions]

Internet telephones use the Internet Protocol (IP) to deliver voice and other multimedia in real time between two or more people over a network. Internet telephones appeared in the mid-1990s with the introduction of Internet telephone software. Internet telephone software is designed to run on your computer, along with voice cards, speakers, microphones and modems. The software compresses the voice signal and converts it into an IP packet for transmission over the Internet. However, a basic PC-to-PC internet phone can only be used when both are using internet phone software. [0003]

Internet telephones offer the opportunity to design a global multimedia communication system that can ultimately replace the foundation of existing circuit exchange telephones. In a relatively short period of time. Internet telephones have made rapid progress. Many software development companies now offer PC phone software. [0004]

Internet telephones are relatively connection-based sessions. Generally speaking, the first internet protocol is used to establish a session and discuss the possibility of that session, and the second internet protocol is used to carry the actual media over an IP network. Will be done. With the proliferation of PC-based Internet telephone software, it has become necessary to standardize protocols. Within the IP network, one current method for handling calls and setting up and decomposing sessions is Session Initiation Protocol (SIP). The SIP protocol was designed to handle calls between devices such as PCs that are directly connected to IP-based networks. [0005]

IP phones benefit both users and data distributors in terms of cost and variety of media, so they are effectively introduced on the basis of traditional networks provided by the Public Switched Telephone Network (PSTN). Moreover, in addition to its extensive nature, the PSTN offers a rich set of telephone services. Therefore, there is a desire to integrate the PSTN with IP networks, including the Internet and private intranets. Therefore, a call initiated by a phone on the PSTN may eventually need to be delivered to a second phone on the PSTN over an IP-based network. For this to work properly, all calls regarding signal information must pass through the IP network without any loss of information. [0006]

The SIP protocol is sufficient to handle the setup, connection and disconnection of most calls with respect to signals. However, the SIP protocol does not deal with how to carry signal information during a call. An example of signal information during a call is the suspension and resumption of ISDN-defined processing. [0007]

In addition to the signals during an ISUP call, the second type of telephone session control information that needs to be carried during a session is the information created by DTMF or "dial plus". Today, there are various telephone services that require the use of DTMF digits. To design these features, DTMF information needs to be carried both as part of the media stream or Real-Time Transport Protocol (RTP) flow and as part of the signal or control path. However, in the IP phone, the media and the control path are separated. In an IP environment, intellectual services are provided by the SIP proxy server, which is in the control path, not in the media path. Therefore, SIP proxy servers that make IP networks intelligent do not pay attention to DTMF information. [0008]

[Means for solving problems]

The present invention provides methods and systems for a transport public switched telephone network that terminates signals through an Internet Protocol (IP) network. The system of the present invention includes a gateway between the PSTN and the IP network. The gateway according to the present invention is a signaling system 7 (Signaling System) from PSTN. Whether to receive a telephone signal message such as 7) or Integrated Services Digital Network (ISDN) or User Part (ISUP) message and map the telephone signal message to an Internet Protocol signal message such as Session Initiation Protocol (SIP) message. Decide whether or not. If the telephone signal message does not map to an Internet Protocol signal message, the gateway packages the telephone signal message into an Internet Signal special message for transport over the IP network. If the gateway of the present invention receives an Internet Protocol signal special message, the gateway unpackages the telephone signal message from the Internet Protocol signal special message to carry over the public switched telephone network. [0009]

The gateways of the present invention are particularly adapted to transfer PSTN in-call signals over IP networks. The gateways of the present invention are also adapted for transporting DTMF signals over IP networks. When the gateway receives the DTMF signal from the PSTN, the gateway converts the DTMF signal into a digit and packages the digit in an Internet Protocol signal special message for transport over the IP network. The gateway also packages DTMF signals into IP media transport protocol messages that transport over IP networks, such as Real-time Transport Protocol (RTP). [0010]

BEST MODE FOR CARRYING OUT THE INVENTION

FIG. 1 is a block diagram of remote communication according to the present invention. FIG. 2 is a block diagram showing the gateway of the present invention. FIG. 3 is a block diagram of the signal inlet gateway according to the present invention. FIG. 4 is a block diagram of the signal exit gateway according to the present invention. FIG. 5 is a flowchart of the media entrance gateway according to the present invention. FIG. 6 is a call flow diagram illustrating the method of the present invention and the operation of the system. [0011]

First, with reference to the drawing of FIG. 1, the remote communication system is broadly indicated by reference numeral 11. System 11 includes IP networks broadly referred to as 13 and various types of traditional telephone networks, including a first PSTN 15 and a second PSTN 17. The traditional telephone network can also include the personal telephone network 19. The conventional telephone network is suitable for providing a conventional telephone service and an interface with the IP network 13 according to the present invention. [0012]

PSTN 15 and 17 are dedicated to providing traditional telephone subscriber-to-subscriber telephone services. The telephone 21 is operably connected to initiate the switch 23, thereby accessing the telephone network broadly indicated by 25. The call is sent through network 25 to termination switch 27. According to the present invention, the termination switch 27 is operably connected to the PSTN / IP gateway 29. The operation and structure of the PSTN / IP gateway 29 will be described in detail below. [0013]

The personal telephone network 29 is dedicated to providing telephone services to members of an organization in a particular area. The user uses the telephone 31 to make a call to another member of the organization or an external party. The telephone 31 is operably connected to a private branch exchange (PBX) 33. The PBX33 is operably connected to the corporate gateway 35 and is similar in structure and operation to that of the PSTN / IP gateway 29. The corporate gateway 35 is operably connected to the personal intranet 37, which in turn is connected to the IP network 13. Intranet 37 is also dedicated to providing IP phone services to IP-capable PCs 39 and IP phones 41. [0014]

Subscribers using telephones 21 and 31 can talk to each other and to other telephones connected to system 11, or to Internet devices such as personal computers 39 and IP phones 41. Such calls are set up and sent using a combination of standard telephone and Internet signals and media transport protocols. Intellectual network services can be provided by proxy server 42 within the PSTN or IP network. [0015]

The ISDN User Part (ISUP) defines protocols and messages for establishing and destroying voice and data through the public switched telephone network. Network elements use ISUP messages and their parameters to configure and break voice and data circuits. The basic messages used in connections that set up or break voice circuits are the initial address message (IAM), the address completion message (ACM), the response message (ANM), the release message (REL), and the release circuit message ( RLC). [0016]

The IAM begins to occupy the outgoing circuit and is sent forward by the switch to transmit numbers and other information about the path and handling of the call. The ACM message is sent backwards to indicate that all the address signals needed to make the call have been received by the termination switch. ANM is sent backwards to indicate that the call has been answered. After the ANM is received, the call proceeds. The REL message is sent in both directions to indicate that the circuit has been released due to the reason or cause of the message. The RLC message is sent while the REL message is received when the appropriate circuit is idled. [0017]

Session Initiation Protocol (SIP) is a standard proposed by the Internet Engineering Task Force (EITF) for establishing, modifying, and terminating multimedia IP sessions. An IP phone call is a special multimedia session in which voice data is exchanged between parties. SIP is a customer / server protocol in which a customer makes a request and a server responds. SIP now defines six requirements or methods, including INVITE, ACK, OPTIONS, REGISTER, CANCEL, BYE. [0018]

The INVITE request is used to ask if there is a party called in the multimedia. The ACK method is sent to allow a new connection. OPTION is used to get information about the capabilities of the server. In response to the OPTION request, the server returns the method it is assisting. The REGISTER method tells the server where the user is. The CANCEL method ends parallel searches. You can try several places when the server is trying to reach you. When the user is reached, the rest of the search can be cancelled. The customer sends the BYE method to end the session, and for a two-party call, the BYE method ends the call. [0019]

When the server receives the request, it sends back a response. In SIP, the response type is defined by a 3-digit code number. There are six main types of responses, including 1XX notifications, 2XX successes, 3XX transfers, 4XX request failures, 5XX server failures, and 6XX overall failures. An example of a notification response is a 180 ringing response. An example of a successful response is a 200 OK response. [0020]

The interface between the standard telephone section of system 11 and the IP section of system 11 is provided by gateway 29 and gateway 35. Gateway 29 and gateway 35 provide protocol conversion for both signals and media. Refer to Figure 2 to illustrate the architecture and services provided by gateways such as PSTN / IP gateway 29 and gateway 35. In Figure 2, the ingress gateway is indicated by 43 and the exit gateway is indicated by 45. The "entrance" and "exit" labels are used purely for identification purposes and should not be construed in a limited sense. Thus, one of ordinary skill in the art will appreciate that when a call is initiated at the opposite end of system 11, gateway 43 can act as an ingress gateway and gateway 45 can act as an exit gateway. The ingress gateway 43 includes a signal gateway 47 and a media gateway 49. Similarly, the exit gateway 45 includes a signal gateway 51 and a media gateway 53. [0021] [0021]

The signal gateway 47 is connected between the telephone signal link 56 and the IP network 55. The signal link 56 is connected to the signal transfer point (STP) 57, which in turn is connected to the switch 59. In an embodiment of the invention, the signal gateway 47 provides a bidirectional protocol translation between the ISUP message and the Session Initiation Protocol (SIP) request and response, which is selectively referred to as the message. The media gateway 49 is connected between the voice trunk 61 and the IP network 55. The voice trunk 61 is connected to the switch 59. Media gateway 49 provides conversion between standard time division multiplexing (TDM) voice circuits and IP media transport protocols such as Real Time Transport Protocol (RTP). [0022]

Similarly, the signal gateway 51 is connected to the IP network 55 and the telephone signal link 63. Signal link 63 is connected to STP65, which in turn is connected to termination switch 67. The media exit gateway 53 is connected between the IP network 55 and the voice trunk 69. The voice trunk 69 is connected to the termination switch 67. [0023]

Mappings between PSTN media and signal transfer and between IP media and signal transfer are illustrated with respect to the call flow diagram of FIG. FIG. 6 shows the progress of a call between the outgoing switch 59 and the terminating switch 67 of FIG. Communication of signals between the switch and the associated STP is omitted for clarity. Communication of PSTN signals is done through ISUP messages. PSTN media transfer is done through TDM. The IP portion of the call is carried between the entry gateway 43 and the exit gateway 45. Communication of IP signals is done through SIP, and media transfer is done through RTP. [0024]

The call begins with an IAM message sent from the outgoing switch 61 to the entry gateway 43. The ingress gateway 43 translates the IAM message into a SIP INVITE message, which is sent to the egress gateway 45. The exit gateway 45 translates the INVITE message into an IAM that sends it to the termination switch 67. In response to the IAM message, the termination switch 67 sends the ACM message back to the exit gateway 45. The egress gateway 45 translates the ACM message into a SIP180 ringing response addressed to the ingress gateway 43. Ingress gateway 43 translates the 180 ringing response into an ACM message. When the called party answers the call, the termination switch 67 sends an ACM message to the exit gateway 45, which translates it into a 200OK response. When the ingress gateway 43 receives the 200 OK response, it sends an ANM to the outgoing switch 61 to establish the call. [0025]

During a call, the media gateways of the ingress gateway 43 and the exit gateway 45 perform bidirectional conversion between TDM and RTP. During a call, there may be communication of signals that are not supported by the current execution of SIP, either by system-generated ISUP messages or by user-generated DTMF digits. Therefore, the present invention provides a special new SIP requirement or method referred to as the INFO method. The INFO method uses, as an argument, all ISUP messages or sequences of dialed digits that do not map to a defined SIP request or response. For example, if the called party is temporarily disconnected, termination switch 67 sends an ISUP stop (SUS) message to the exit gateway 45. Thus, according to the present invention, the exit gateway 45 packages the SUS message in the INFO message for transport to the entry gateway 43. The ingress gateway 43 unpacks the INFO message and sends the SUS message to the outgoing switch 61. When the called party is reconnected, termination switch 67 sends an ISUP stop message to exit gateway 45. Once again, RES messages do not map to any predefined SIP request or response. Thus, according to the present invention, the exit gateway 45 packages the RES message in the INFO message for transport to the exit gateway 43. The ingress gateway 43 unpacks the INFO message and sends the RES message to the outgoing switch 61. [0026]

During a single call, the user can initiate signal communication by entering a DTMF digit. DTMF digits are intended to be used by termination equipment such as response machines that return messages. As an alternative, DTMF signals are considered to be used by elements of intelligent networks and may be part of a PSTN such as a SIP proxy server or elements of an IP network. Due to the separation of signal communication and media transfer between the IP parts of a call, it is necessary to send DTMF information in both the signal communication path and the media path. [0027]

Again, referring to FIG. 6, the TDMF signal tone is transmitted from the transmit switch 61 to the inlet gateway 43 in the DTM stream. The DTMF signal sound is received by the media gateway 49 of the entrance gateway 43. Therefore, the media gateway 49 executes two processes at the same time. The media gateway 49 converts DTMF signal tones into digits and passes those digits to the signal gateway 47 of the ingress gateway 43. The media gateway 49 also converts DTMF signal tones into RTP packets for transport to the media gateway 53 of the ingress gateway 45. The signal gateway 47 of the ingress gateway 43 packages the digit received from the media gateway 49 in the SIP INFO request for transport to the signal gateway 51 of the exit gateway 45. INFO messages with packaged digits can be used by any of the SIP proxy servers in the IP signal path that provides intelligent network services. The exit gateway 45 ignores the digitized INFO request and translates the RTP packet into a TDM signal. [0028]

When the call is complete, one of the parties will stop. In the example of FIG. 6, the called party ends the call and the termination switch 67 sends a REL message to the exit gateway 45. The egress gateway 45 translates the REL message into a BYE request, which is forwarded to the ingress gateway 43. The ingress gateway 45 translates the BYE request into a REL message to the outgoing switch 61. In response to the REL message, the outgoing switch 61 returns the RLC message to the ingress gateway 43. The ingress gateway 43 sends a 200 OK response to the exit gateway 45. The egress gateway 45 returns the SIP ACK request to the ingress gateway 43 and the ISUP RLC message back to the termination switch 67. [0029]

The processing of the gateway according to the present invention is shown in FIGS. 3 to 5. First referring to FIG. 3, the inlet signal gateway processing is illustrated here, and in step 71, the inlet signal gateway determines whether the signal is a digit from the media gateway when the signal is received. Decide. If so, the ingress signal gateway packages the digit as a SIP INFO request in stage 73. If not, the entry signal gateway decides in decision stage 75 whether the message maps to a SIP request or response. If not, the ingress signal gateway packages the signal in the SIP INFO message in step 77. At decision stage 75, if the signal maps to SIP, the ingress signal gateway performs the appropriate map at step 79. [0030]

Refer to Figure 4 to show exit gateway processing. When the exit gateway receives the signal, the exit gateway determines in decision stage 81 whether the signal is a SIP INFO request. If not, the exit signal gateway makes an appropriate map at step 83. If the signal is a SIP INFO request in decision phase 81, the exit signal gateway determines in decision step 85 whether the argument of the request is a digit. If so, the exit dimension gateway ignores the digit as shown in step 87. At decision stage 85, if the request argument is not a digit, the exit signal gateway unpacks the ISUP message from the SIP INFO request at stage 89. [0031]

Refer to Figure 5 to show entry media gateway processing. The ingress media gateway receives the TDM data and determines in decision stage 91 whether the data is a DTMF signal tone. If not, at stage 93, the ingress media gateway packages the incoming data into RTP packets. At decision stage 91, if the incoming data is DTMF signal tones, the ingress media gateway packages the signal tones into RTP packets, converts the DTMF signal tones into digits, and converts those digits at step 95. At stage 97, it is sent to the signal gateway. [0032]

As described above, the present invention provides methods and systems that assist in communicating signals during a call for PSTN termination calls that traverse an IP network. Those skilled in the art will recall alternative embodiments that have the advantages of the present invention. Therefore, the above is for the purpose of explanation and not for the purpose of setting limitations.

[Simple explanation of drawings]

[Figure 1]

It is a block diagram of the remote communication by this invention. [Figure 2]

It is a block diagram which showed the gateway of this invention. [Fig. 3]

It is a block diagram of the signal entrance gateway by this invention. [Fig. 4]

It is a block diagram of the signal outlet gateway by this invention. [Fig. 5]

It is a flowchart of the media entrance gateway by this invention. [Fig. 6]

It is a call flow chart which illustrates the operation of the method and the system of this invention. [Explanation of symbols]

13,55 ... IP network 15 ... 1st PSTN 17 ... 2nd PSTN 19 ... Personal Phone Network 21a, 21b, 31 ... Phone 23a, 23b, 27a, 27b, 59, 67 ... switch 25a, 25b ... PSTN 29a, 29b ... PSTN / IP gateway 33 ... PBX 42 ... SIP server 43,45 ... gateway 47,51 ... Signal gateway 49,53 ... Media Gateway 57,65 ... STP