TW201036378A - Voice gateway architecture and its communication method - Google Patents

Abstract

The present invention relates to voice gateway architecture and its communication method. The voice gateway architecture comprises at least one gateway, a network hub and a plurality of internet phone devices. The gateway is the network switch connected to the network hub for providing internet voice services, and is used to integrate keys by means of program call so as to have functions of establishment and transmission of internet voice and management of key encryption and decryption to centrally manage the operations of network calls. The communication method is to use the gateway to identify the type of calls and format of voice packets for executing internet voice transmission operations. Therefore, the present invention can ensure the establishment of the call and significantly reduce the probability of call failure.

Description

201036378 VI. Description of the Invention: - Technical Field of the Invention The present invention relates to a voice gateway architecture and a communication method thereof, and more particularly to an IP network voice transmission technology using a Session Initiation Protocol (SIP). (Voice over Internet Protocol, VoIP) voice gateway architecture and its communication method. [Prior Art] With the increasing application of fiber-optic networks, the cost of fiber-optic construction is becoming cheaper and cheaper. At the same time, applications using fiber-optic high-speed transmission are relatively more and more. Among them, the network The telephone is a product that uses the Internet and provides voice calls through Voice over Internet Protocol (VoIP). It has low call cost, low cost of construction, and convenient maintenance. Therefore, the more More and more companies use it for their own phone calls within their own companies, especially for business calls between multinational branches, to significantly reduce the cost of international calls. In general, Internet telephony encodes voice into a User Datagram Protocol (UDP) packet that supports the Real-time Transport Protocol (RTP), and then directly uses the Internet to access the IP network. The voice transmission technology is transmitted to the other party's telephone device, and the call device can be called after the other party's telephone device is decoded. In other words, the two parties only need to have the network telephone device supporting the IP network voice transmission technology at the same time, for example, a computer or a palmtop computer. Or a mobile phone, you can make calls over the Internet. Among them, IP network voice delivery technology is based on Session Initiation Protocol (SIP) 3 201036378

The way to establish a call is adopted by various vendors. The protocol uses a SIP 'server to start, manage, and suspend the voice of the network voice by TCP/IP and encode the message using HTTP/1 · 1. The transmission is performed in such a manner that after the sender/reception terminal registers with the SIP server to obtain the SIP service, the packet transmitted by the call can be read without a complicated decoding process. However, with the increasing number of cybercrime cases, Internet security kings are getting more and more people's attention, so there are manufacturers of related Internet telephony through key exchange. The purpose of the encryption and decryption call is reached. However, the way of encrypting and decrypting the VoIP calls of different manufacturers is not the same. Therefore, if one of the VoIP devices does not support the other party's Garda technology, it cannot establish security with the other party. The call, even the failure to dial and the call failed, thereby limiting the ease of use of the Internet phone, which is a long-standing incompatibility of the Internet phone. In addition, the 'encrypted call record has the problem of tracking difficult and unmonitored questions, which leads to a dead angle in public security prevention, making sensitive units (such as the emotional unit) or countries with more open information (such as China). The mainland government) adopts a blockade or restricts the use of encrypted VoIP for the function of VoIP. The goal of R&D therefore is 'how to make VoIP calls securely in encrypted form, and to allow VoIP calls to be recorded and stored to prevent VoIP from being abused by lawbreakers. It has been a long time since the relevant vendors have In the light of this, the present inventors have actively conducted research and improvement in order to solve the problems in the use of the above-mentioned conventional Internet telephones, and have finally developed after continuous trial and effort. this invention. The first object of the present invention is to provide a voice gateway architecture application-gateway system-to-management area for voice transmission, so that various network telephone devices can transmit voice calls through the gateway to avoid Any situation where the encryption or decryption method or the key mismatch causes the call to fail. A second object of the present invention is to provide a voice gateway architecture, which is to establish a perfect call quality by processing a signal and voice packet of an application session initiation protocol (Session Initiati〇n Pr〇t〇c〇1, sip). It is a problem that the traditional VoIP phone fails due to the software or hardware in the device does not support the encrypted call. A third object of the present invention is to provide a voice gateway architecture for a gateway having a recordable call content, which stores data of a call record through a database, so that the network phone can be encrypted in use. To ensure the safety of the call, but also to prevent illegal use (such as: trade secrets leaked) or as a record of the post-crime. For the above purposes, the voice gateway architecture of the present invention includes: a network sharer (Hub) connected to the Internet; a plurality of VoIP-enabled network telephone devices, which can be wired or wireless. The way to connect to the Internet through the network sharer;

A gateway device is a service that is connected to the network sharer (A 201036378 private branch exchange, PBX) to provide voice calls for the network, and uses a program call to integrate the keys. And has a recording function, to store the call record; a database 'used to store the call record recorded by the gateway. Therefore, with the above architecture, when a network telephone device wants to make a call with the corresponding network telephone device, the SIP server in the Internet must be registered through the network share device to obtain the session initiation. The protocol (SIP) service is registered with the gateway through the network sharer, and the gateway device registers with the SIP server to obtain the SIP service through the network sharer, and then passes the gate. Through the network sharer, the tracker uniformly performs voice transmission and establishes a call with the SIP server or the corresponding network telephone device. A fourth object of the present invention is to provide a voice gateway communication method, which is a method for establishing a call and transmitting a voice packet by using the voice gateway architecture described above, to solve different types of network telephone devices due to incompatibility between software and hardware. The problem of the call failed. Therefore, for the above purposes, the voice gateway communication method of the ten invention includes: establishing a call step: when the gateway receives a SIP call signal when a call end establishes a call with the intended call end The gateway device establishes a call according to the type of the call to be established and the type of the network telephone device at the receiving end, wherein the call type can be divided into an encrypted or unencrypted call; the packet processing step: when the gate When receiving the voice packet sent by the sender, the tracker transmits the π a packet to the receiver according to the call type and the format of the voice packet, so as to ensure that the receiver can voice seal 6 201036378 The action of decrypting the packet. Therefore, by the above method, the gateway can be used to confirm the call and the format of the packet for the voice transmission of the network voice, thereby ensuring the establishment of the call, and greatly reducing the probability of the call failure of the network telephone device. . DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS In order to make those skilled in the art understand the present invention, the preferred embodiments of the present invention will be described in detail below with reference to the drawings. Referring to the first figure, the voice gateway architecture of the present invention is set under the network communication environment of the SIP_base network protocol, which includes: a network knife (Hub) (2) ' Connected to an Internet (8); a plurality of VoIP-enabled VoIP devices (3) connected to the network (8) via a network sharer (2) via wire or wireless The gateway (1) is a kind of network switch (PbX) connected to the network sharer (2), which is used to provide a network voice call service and utilizes a program call. The way to integrate the gold input, and has the function of monitoring the call to store the call record; the database (1 〇) is used to store the call record stored by the gateway device. Therefore, 'by the above-mentioned architecture', when a network telephone device (3) wants to connect with the word 'speaking network phone 奘蒈> C C 3 ), it must pass through the network knife (2) ) to the Internet (8) SIp server (4) =. The main book is used to learn the session initiation protocol (sip) service, and then through the network sharer (2) to register the gateway (work) with the gateway device (丄) and then through the network share device (2) ) Registering the sip server (4) with the 7 201036378 = Π > service, and then passing the channel (1) through the network 2: one with the SIP feeder (4) or the corresponding network (31) ) Voice transmission and call setup. The above-mentioned call record includes: - CallDetailRec〇rd, at least the end address, the receiving end M, the st, the call start time, the call end time and the call duration; ^ Ο ❹ The call recording is a call file between the two parties during the call, in which the voice is set up by Ann & the sound is recorded, and the time of the call record is set to the file name. The above-mentioned gateway device (1) further includes: a voice processing unit that can compress a large voice broadcast into a :: case, so that the database (1.) can store more calls. The voice processing unit can be a sound (four) soft body (for example:

Alntan MP3 Enc〇der (LAME) software), large recordings of Dehong 厥 # 槽 较 较 较 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩 压缩The database (the eighth) can store the recording files of more calls. To help the reviewing committee understand the voice gateway architecture of the present invention, the voice closed architecture of the present invention is applied to a network in a commercial building. The commercial building includes a plurality of telephone devices having a function of v〇Ip (for example, a car, a mobile phone, a laptop, a PDA, etc.), and the Internet phone is placed through a network installed inside the building. After the road sharer is connected to the Internet 201036378, you can make an Internet call. 凊Refer to the second figure, when a network telephone device (g) is from the external Internet, the road (not in the commercial building) When the network telephone device (3) is to be dialed to the internal building of the commercial building, the network telephone device (9) sends a sip invitation signal and obtains a session initiation agreement through a SIP server (4). (SIp) After the service is obtained, the SIP server (4) sends the Xiao SIP request signal to the corresponding network telephone device (3)' to establish the relationship between the two network telephone devices (9) (3). The call 'the network telephone device (3) receives the SIP request signal, the gateway device (1) will first receive the SIP request message and process the sip invite signal, so all the snM server ( 4) The transmitted/received signal will be transmitted to the gateway (work) first, and the gateway (丄) will be used to make a call with the sip feeder (4) or the corresponding network telephone device (3). The establishment and voice transmission. Therefore, the VoIP device (3) in the commercial building transmits the VoIP call through the gateway (1) and the SIp server (4) in the Internet. And transmission, in addition, the data of the call record can be stored through the database of the gateway (^), so that the use of the Internet phone can prevent illegal use (for example, leakage of trade secrets) or as a crime Record of evidence. The voice gateway communication method of the present invention applies the above voice gate The channel architecture - to carry out, includes: = establishing a call step. When the gateway receives a call from the sender and the corresponding SIP invites the clear signal, the caller will establish a call according to the caller. The type and the type of the network telephone device on the receiving end determine the type of call corresponding to 9 201036378. In Jt, the processing step of the spinning packet is encrypted or unencrypted; when the packet is received, the voice sent by the calling terminal is received. The voice packet is transmitted to the received (10): packet format, and the corresponding packet is decrypted. Knowing 'to ensure that the receiving end can seal the voice to the above communication method by using the gateway to confirm the format of the packet to perform the network Lulu has confirmed that the type of call and the Ο Ο are established, which greatly reduces the transmission operation, so it can ensure the probability of a call failure. Please refer to the third figure, the upper and lower steps: erect the call step, including the step A-1 (i Ω, when the number of the device is opened * the establishment of the: the channel device receives the SIP message Step A-2 (1. °9 The conversation between the terminal and the receiving end; in the same regional network, judging whether the receiving end and the interrogator are not, proceed to step A_3 (1〇3) if 'Step A-4 (1 〇 4); Step A-3 (1 〇 is encrypted or not added, 1) The type of call that wants to establish a call is reversed, then enter the step call, and if yes, enter (4) A_5 ( 1 〇 5), Step Μ1〇(.1〇6); Whether it is encrypted 4). Determine if the type of call the caller wants to establish is no; go to step t 'If yes' then go to Step A-7 (1 〇7) ), if step A-5 (1 〇 to decrypt and t 5): the gateway to the encrypted SIP invitation 0 9); After recording the encryption key of the signal, go to step Α·9 (1 step Α-6 (丄η 6). The gateway establishes an RTp 201036378 call with the called end; y step A 7 (1 0 7): the gateway solves the encrypted SIp invite signal, and already After the encrypted record of the signal, enter step Α8 (1 0 8); Step A-8 (1 π Q, . 丄υ 8). The gateway forwards the SIP invite signal to The receiving end; Ο Ο Step A9 (1 〇 9): The gateway determines whether the receiving end supports SRTP (Queen Instant Transmission Agreement (e Ding ^^ Qing (10)

ProtocoUSRTP) to transmit) If the security call is established, go to step A_1〇(1 1 〇)' If no, Bay 1J enters step A-ll (1 1 1 ); Step A_1〇(1 1 0 ): The gate The router establishes a secure call with the SRTp to the called end; Step A-ll (1 i丄): The gateway establishes RTp with the called end (transmits by Real-time Transport Protocol (RTP)) ) the call. Please refer to the fifth figure to help the review committee understand the voice gateway communication method of the present invention. Now, a caller (5 1) supporting SRTP calls and a caller (54) supporting only RTP calls are to be established. SRTp calls are described as an example. Firstly, the utterer (51) is a VoIP device that is different from the gateway (1) and belongs to the local area network. 1) The receiving end (54) belonging to the same local area network, wherein the calling end (51) and the receiving end (54) establish a call system sequentially including the following procedures: 11 201036378 Procedure one (500): the calling end (51) Send a sip invite 'signal to a SIP server (4) and register with the server (4) to get a SIP service; program two (5 0 1): the sip server (4) will Before the SIp invitation signal is transmitted to the receiver (54), the gateway (1} receives the sip invitation signal and determines whether the receiver (5 4) supports the sRTp call; Procedure 2 (502): The receiving end (54) cannot support the Srtp 〇 call. Therefore, the gateway (1) will unlock the SIP ping signal sent by the utterer (51) under the SRTP call and change to the receiving end (5 4 Establishing an RTP call, that is, the receiving end (5 4 ) receives a SIP invitation signal in an RTP call format sent by the gateway (1); Sequence 4 (503): The receiver (54) replies to the gateway with an acknowledgment signal in the -RTP format (1 program five (5 0 4): the gateway (" again from the receiver ( 5 4) The confirmation signal sent is changed to the SRTP call format confirmation signal transmission ◎ to the SIP server (4); program six (5 0 5) · The SHM server (4) is transferred to the SRTp The confirmation signal of the call format is sent to the calling terminal (5 i ). At this time, the calling terminal (5 1 ) and the receiving terminal (5 4 ) complete the establishment of the call. Therefore, the present embodiment uses the gateway device. ) and the sender (5 workers) • Establish a srtp call (5 〇 6 ) and RTp call (5 〇 7 ) respectively with the receiver (5 4 ) 'Successfully solved the past sender (5 丄) To establish SRTP The call and the call end (54) only support the call failure caused by the RTP call. 12 201036378 As shown in the fourth figure, the above packet processing steps include the following steps: , . ) '* The gateway Receive the voice of the caller, 匕 '' start the voice packet transmission operation; Step Β-2 Γ ο /^ 〇\ 于士门〇2 · · Determine whether the receiver and the gateway belong to the same regional network. If the county is on the right, go to step B_3 ( 2 〇 3 ). Otherwise, go to step B-4 ( 2 〇 4 );

G ^ 7 (2〇3). If it is judged whether the voice packet is pushed into the SRTP call, the process proceeds to step ^"2〇5). If not, the process proceeds to step 匕6 (2〇6); = 3-4 (2 0 4): determine whether the voice packet is a "new" pass packet, and if so, proceed to step B7 (2 () 7), if not, go to step B-8 (2 0 8); ^ step B_5 ( 2 0 5 ) ·· The gateway unlocks the SRTp call format packet, and then compiles it into a voice packet of the RTp call format, and then proceeds to step B-9 (2 0 9 ); Step B-6 ( 2 DR . a female # ^ϋ6)·Send the voice packet to the receiving end; Voice Seal 0: Set the gateway to unlock the SRTP call format. The day packet is encoded into the SRTP call supported by the caller. Formatted voice packet, enter step b_13 (2 i 3 grid / 8 ( 2 Q 8 ) · Because the voice packet is a non-SRTP call packet' (4) The gateway will break the call according to the decision established by the call session ^ No encryption is required, if yes, enter 2 (212), right no, then go to step B-13 ( 2 1 3 ). Step B: 09): Determine whether the receiving end has received sRTp 13 201036378 call format deduction Packet capability If yes, go to step β·1〇( 2 1 〇) 'If not, go to step Β·η ( 2 1 1 ); y 〇 (21〇): the gateway encodes the voice packet into SRTP call The voice packet of the format is transmitted to the receiving end to complete the transmission of the voice packet; Step B-11 (2 11): the gateway device directly transmits the voice packet of the RTP call format to the receiving end, that is, completes the voice packet. Transmitting; Step B-12 (2 1 2): The gateway encodes the voice packet into a voice packet of the SRTP call format supported by the dual-end terminal, and proceeds to step b_13 (213); Step B-13 (2 1 3): The voice packet is transmitted to the receiver to complete the transmission of the voice packet. According to the above method, when a gateway receives a voice packet sent by a terminal, it will transmit the corresponding voice packet to a receiver according to the type of the call and the format of the voice packet. Therefore, the gate is utilized. The device can confirm the type of the call and the format of the packet for the transmission of the network voice, which ensures the establishment of the call and greatly reduces the probability of the call failure of the network telephone device. However, although the present invention has been disclosed in the above preferred embodiments, it is not intended to limit the present invention. The voice gateway architecture can also be applied to a wide variety of network environments, but any skilled person is familiar with the art. The scope of patent protection of the present invention is defined by the scope of the patent application attached to the specification, without departing from the spirit and scope of the invention.

For the reference D [Simplified description of the drawing] 201036378 The first figure is a schematic diagram of the voice gateway architecture of the present invention. The second circle is a schematic diagram of the network telephone transmission wheel of the present invention. The third figure is a flow chart of the steps of establishing a call according to the present invention. The fourth figure is a flow chart of the packet processing steps of the present invention. The fifth figure is a schematic diagram of the present invention for establishing an SRTP call by a caller supporting an SRTP call and a caller supporting only RTP. [Main component symbol description]

(1) Gateway (2) Network sharer (3) Internet phone device (9) Internet phone device (4) SIP server (10 1) Step A-1 (10 2) Step A-2 ( 10 3) Step A-3 (10 4) Step A-4 (10 5) Step A-5 (10 6) Step A-6 (10 7) Step A-7 (10 8) Step A-8 (10 9 Step A-9 (110) Step A-10 (111) Step A-11 (201) Step Bi 15 201036378 ( 2 0 2 ) Step B-2 ' ( 2 0 3 ) Step B-3 - ( 2 0 4 Step B-4 ( 2 0 5 ) Step B-5 ( 2 0 6 ) Step B-6 ( 2 0 7 ) Step B-7 ( 2 0 8 ) Step B-8 ( 2 0 9 ) Step B-9 Ο (210) Step B-10 (211) Step B-11 (212) Step B-12 (213) Step B-13 (51) Talker (54) Receiver (5 0 0) Procedure One (501) Procedure Two Ο ( 5 0 2 ) Program three ( 5 0 3 ) Program four ( 5 0 4 ) Program five ( 5 0 5 ) Program six ( 5 0 6 ) SRTP call. ( 5 0 7 ) RTP call 16

Claims (1)

201036378 VII. Patent application scope · 1 A voice gateway architecture is set up in a siP-base network • Dragon protocol environment, which includes: Network/7 sub-devices, connected to the Internet a plurality of VoIP-enabled VoIP devices that can be connected to the Internet via a wired or wireless means: a gateway, a network connected to the network sharer The switch ^ is used to provide the service of the voice call of the network, and uses the function of the program to call ^ & Jin Yu' and has the function of monitoring the call to store the call record. 2. The voice inter-channel architecture as described in claim i, wherein the call record includes: a connection record, which includes at least a caller address, a callee address, a call access channel, and a call start. Time, call end time and call performance time; the case call recording is to record the voice of both parties during the call into a file. 3. As described in the second paragraph of the patent scope, the voice closed circuit structure, in the gateway The device further includes: ^ a database for storing the call records stored by the gateway; - a voice processing unit for compressing the sound files. 4. The voice gateway structure as described in claim 3, wherein the voice processing unit is an audio compression software ', '5, a voice communication method, and the patent application scope is the first item. The voice gateway architecture includes: a receiving end establishing a calling step: when the gateway receives a sending sleeve..._ 17 201036378 When a SIP invitation signal is sent when a call is to be established, the calling message is sent according to the The type of the call to be established and the type of the network telephone device of the called end determine the type of the call corresponding to each other. In the middle, the call type is encrypted and non-encrypted; the packet processing step: when the gateway receives the call When the voice packet sent by the originating end is transmitted, the voice packet is transmitted to the receiving end according to the type of the call and the format of the voice packet. Ο 〇 6. For example, the voice gateway communication method described in item 5 of the patent application, wherein the type of the secret telephone is securely transmitted in real time, Real-tune Transp〇rt Pr〇t〇c 〇 1, SRTp ) to transmit. 7 * The voice gateway communication method according to claim 5, wherein the non-encrypted call type is transmitted by an instant transmission protocol ("Transp〇rt Pr〇t〇c〇i, RTp"). The voice gateway communication method according to claim 5, wherein the establishing the call step comprises:, step A1. When the gateway receives the SIp request signal, the gate is The tracker starts to establish the call between the caller and the caller; Step A_2··determines whether the callee and the gateway belong to the same regional network,: Si θ α , w , 疋, then enter step A-3, if If no, go to step Ad; A3. Determine whether the call type of the call is to be encrypted or if the call is 'if yes', then go to step A-5; otherwise, go to step A-6; Judging whether the type of the call to be established by the calling terminal is plus, if the message is 疋, then proceeding to step A-7, and if not, proceeding to step A-8; step A-5. The gateway is paired with the encrypted sip After the signal is decrypted 18 201036378 and the encrypted record of the signal is recorded, 'Enter step A-9; Step A-6: The gateway establishes an unencrypted call with the called terminal. Step A-7: The gateway After decrypting the encrypted SIP invite signal and recording the encryption key of the signal, go to step A-8. Step A-8: The gateway transmits the SIP invitation signal to the receiving terminal. Step A-9: The gateway determines whether the receiving end supports the establishment of the encrypted call. If yes, proceed to step A- 10, if no, go to the step Step A-10: The gateway establishes an encrypted call with the called end. Step A-11: The gateway establishes an unencrypted call with the called end. 9. The voice gateway communication method according to claim 5, wherein the 'packet processing step' comprises the following steps: Step B-1: When the gateway receives the voice packet of the calling end, Start the operation of voice packet transmission; #Step B-2: Determine whether the receiving end and the gateway belong to the same regional network, and if yes, proceed to step B-3, otherwise, proceed to step b_4 step Β·3 : Determine whether the voice packet is a packet of the encrypted call format. If yes, proceed to step Β_5. If no, go to step Μ. Step Β-4: Determine whether the voice packet is a packet of the encrypted call format, and if so, Go to step Β_7, if $, go to step Μ. Step Β-5: The gateway unlocks the voice mask of the encrypted call format, encodes it into a voice packet of a non-encrypted call format, and enters / Aixiang B-9; After the child voice packet is sent to the gateway, the gateway is used to unlock the voice mask of the encrypted call format, and then the voice packet is _ into the voice packet supported by the receiver (5) 201036378, and the process proceeds to step B. -13 ; step Step B-8: The looper determines whether the call needs to be encrypted. If yes, go to step B-12, and if no, go to step m. Step B-9: Determine whether the receiver is The ability to have a voice packet, if any, enter the call format into step B-(1) 1 and proceed to step B-1. 'If not, proceed to step Ι-Ι0: the gateway transmits the &seal value of the voice format, its >, and the 曰 packet encoded into an encrypted call format to the receiving terminal. Step Β 11: The gateway transmits the packet; the packet is directly transmitted to the receiving end, that is, the voice packet of the secret call format is transmitted by the voice packet; Step: the gateway encrypts the voice call of the language The voice is encoded into a squad, and the squad is earned into step B_13; 3. The transmission of the voice packet is transmitted. Have to send to the end of the voice to complete the voice seal Eight, the pattern: (such as the next page) G 20