WO2004079994A1

WO2004079994A1 - Ip access system and method for performing voice service and media-stream process

Info

Publication number: WO2004079994A1
Application number: PCT/CN2004/000185
Authority: WO
Inventors: Xuan Luo
Original assignee: Huwaei Technologies Co., Ltd.
Priority date: 2003-03-06
Filing date: 2004-03-08
Publication date: 2004-09-16

Abstract

The present invention discloses a IP access system, comprising: a local exchanger, for performing calling service and controlling resource; an over-access gateway, for connecting subscriber to a wide-band network; an edge trunking gateway, mounted in the IP network, and for performing signaling conversion and media-stream conversion. Said local exchanger connects to said edge trunking gateway over a trunking link, and interacts each other by a narrow-band signaling protocol. Said IP network connects with said edge trunking gateway. The present invention also discloses a IP access method for performing voice service and media-stream process. The system and method according to the invention can be easily perform voice over IP and media service, on the media-stream processing basis. The invention can further improve the quality of speech, reduce the delay in the communication, and simplify the process.

Description

IP access system and method for realizing voice service and media stream processing

The present invention relates to network voice service technologies, and in particular, to an IP access system based on an IP network and a method for implementing voice service and media stream processing. Background of the invention.

With the gradual opening up of the telecommunications market and the formation of a new competition pattern, all-round competition in the communications field is becoming increasingly fierce. Operators build networks that take into account both speed and cost, and gradually shift their investment focus to the most directly user-oriented access layer. With the rapid development of the IP network, it is becoming more and more possible to provide cheap and high-quality voice and data services on the IP network, and it is increasingly receiving the attention of telecommunication equipment manufacturers and operators. For telecommunications operators, the use of existing IP MANs to provide IP voice and multimedia services to users such as IP supermarkets, communities, small businesses, and SOHO is gradually becoming an effective supplementary solution in the construction of access networks.

The existing new operators have the following characteristics when developing voice and multimedia services: 1. The huge cost of local network construction, coupled with the high cost of interconnection and interoperability with the old operators, make them unable to use the local voice services. Competition, so the new operators mainly focus on long-distance services and multimedia services. 2. The bandwidth of the existing IP MAN of the new operator is very rich and has not been effectively used. Therefore, it is required that the voice access to IP and the development of multimedia services best meet the existing network characteristics and the operator's interests.

At present, a similar scheme for implementing multi-service integrated access through an IP network is mainly a next-generation network (NGN) scheme based on softswitch technology. This scheme belongs to a comprehensive, open network architecture that provides voice, data, and multimedia services. NGN uses soft switching technology to separate the functional modules of traditional switches into independent network components. Each component is divided according to corresponding functions and developed independently.

NGN adopts separation of service and call control, separation of call control and bearer technology to achieve open distribution Network structure, making the business independent of the network. Through open protocols and interfaces, services can be provided flexibly and quickly, and individual users can define service characteristics themselves, without having to worry about the network form and terminal types that carry the services.

Although NGN technology has obvious advantages ₃ and is considered the future direction of network construction and development of its new networking solutions can meet the needs of operators. However, from the current situation, there are two main defects:

1. The cost of fully adopting NGN networking is too high, and operators need to purchase a large amount of related equipment to replace and update the existing network.

2. Because it is a new technology, NGN is still being improved, and there are certain risks in large-scale applications.

Therefore, in a comprehensive view, it still takes a certain amount of time for new operators to apply NGN technology on a large scale, which contradicts the goal of new operators who urgently need to launch integrated services. Summary of the Invention

In view of this, an object of the present invention is to provide an IP access system so that it can easily and conveniently implement IP voice and multimedia services.

Another object of the present invention is to provide a method for implementing voice services through IP access, which can meet the needs of a new operator using IP networks to implement voice services based on existing network resources.

Another object of the present invention is to provide a method for processing media streams in an IP network, which can further improve the call quality of the IP access technology, reduce the call delay, and simplify the processing flow.

To achieve the above object, the technical solution of the present invention is implemented as follows:

An IP access system, including an IP network, the system further includes: a local switch for implementing call services and resource control; more than one access gateway for connecting a broadband network and a user terminal to implement IP access; an edge A relay gateway, which is set in an IP network and is used to implement Order conversion and media stream conversion; the local switch is connected to the edge relay gateway through a relay link, and interacts through a narrowband signaling protocol; and the access gateway is connected to the edge relay gateway through an IP network.

The edge relay gateway further includes a loopback control module that controls a media stream to perform a self-loop on the IP side.

In the above solution, the access gateway and the edge relay gateway are media and media gateway control protocols.

(MGCP) link, or H.248 protocol link, or H.323 protocol link, or Session Initiation Protocol (SIP) link. The narrowband signaling protocol includes V5 signaling, or No. 7 signaling, or No. 1 signaling, or PRA signaling protocol. The access gateway is an access media gateway (AMG), or an integrated access device (IAD), or a combination thereof.

The present invention also provides a method for implementing voice services through IP access, including the following steps:

A. The calling user terminal initiates a call to the called user terminal through the access gateway;

B. The edge relay gateway on the calling user terminal and the called user terminal side respectively establishes a real-time transmission protocol (RTP) channel between the edge relay gateway on the two user sides and the access gateway. Protocol format conversion on the local switch side, establishing a connection between the calling user terminal and the called user terminal;

C. The voice service flow between the calling user terminal and the called user terminal is transmitted through the real-time transmission protocol channel of the corresponding side.

In the above process, the step B specifically includes:

B1. The access gateway on the calling user terminal side reports the calling event of the calling user to the edge relay gateway, and the edge relay gateway converts the received call event protocol format and reports it to the local switch; and The relay gateway establishes a real-time transmission protocol (RTP) channel between the edge relay gateway and the access gateway;

B2. The edge relay gateway on the calling user terminal side reports the called user number to the local switch. Only the local switch sends the called user number to the edge relay gateway on the user side to send an instruction about the call event. B3. After the edge relay gateway on the called user terminal converts the indicated protocol format, it is delivered to the access gateway on the called user terminal side, and between the edge relay gateway and the access gateway. Establishing a real-time transmission protocol channel (RTP) channel;

B4 _¾ The access gateway on the called user terminal side notifies the called user terminal of the call event.

Wherein, the step of establishing a real-time transmission protocol (RTP) channel between the edge relay gateway and the access gateway described in step B1 may be performed anywhere before step B6.

In the above solution, a media gateway control protocol (MGCP), or an H.248 protocol, or an H.323 protocol or a session initiation protocol (SIP) format is used between the edge relay gateway and the access gateway; the edge relay gateway The V5 signaling, or No. 7 signaling, or No. 1 signaling, or PRA signaling protocol format is used with the local switch.

In the above solution, in step B2, the edge relay gateway reports the called user number converted by its own protocol to the local switch through signaling, or reports the called user number to the local switch through a real-time transmission protocol (RTP) channel through media streaming. .

In the above solution, the edge relay gateway in step B2 reporting the called user number to the local switch further includes:

B21. The local switch sends a ringback tone to the calling user terminal.

B22. The edge relay gateway converts the pulse code modulation (PCM) ringback tone into an IP packet and transmits it to the access gateway through a real-time transmission protocol (RTP) channel; or instructs the access gateway to send a ringback tone through a control protocol;

B23. The access gateway converts the IP packet into a pulse code modulated ringback tone and sends it to the calling user terminal; or sends the ringback tone to the calling user terminal according to an instruction of the edge relay gateway.

The method further includes: when either the calling user terminal or the called user terminal terminates the service, the RTP channel corresponding to the terminating service party is closed and the related resources are released.

The method further includes: each user completes state initialization and registration in the edge relay gateway and the local switch through the access gateway before accessing. The present invention also provides a method for processing media streams in an IP network. After receiving the call access request initiated by the calling user terminal, the edge relay gateway performs the following steps:

1) judging whether the called user terminal belongs to the edge relay gateway according to the called user information in the call access request;

2) If the called user terminal belongs to the edge relay gateway, use the IP address and port number corresponding to the called user terminal as the peer address on the calling side of the media stream, and use the IP address and port corresponding to the calling user terminal. The number is used as the peer address of the called side of the media stream, and directly connects the media stream between the calling and called parties.

Wherein, the step 2) further includes the following steps:

21) first determine whether the access gateway of the calling and called sides has obtained the RTP address of the media stream receiver;

22) If the RTP address of the media stream receiver has been obtained, the edge relay gateway instructs the access gateway on the calling and called sides to change the peer address on the calling side of the media stream to the IP address corresponding to the called user terminal And the port number, and change the peer address on the called side of the media stream to the IP address and port number corresponding to the calling user terminal, directly connect the media stream between the calling and called parties, and end the current process; if no media stream is obtained The RTP address of the receiver, then proceed to step 23);

23) The edge relay gateway instructs the access gateway on the calling and called sides to set the peer address on the calling side of the media stream to the IP address and port number corresponding to the called user terminal, and The peer address is set to the IP address and port number corresponding to the calling user terminal, and directly connects the media stream between the calling and called parties.

In step 22), the edge relay gateway first cuts off the real-time transmission protocol channel between the edge relay gateway and the access media gateway connected to the calling and called user terminal, and then controls the connection connected to the calling and called user terminal. The ingress gateway changes the peer address on the calling and called sides of the media stream.

In the above solution, after the media streams between the calling and called parties are directly connected in step 2), the method further includes: judging whether a calling or called user terminal on-hook / fork signal or abnormal signal is received. To, the edge relay gateway controls the access to which the calling and called user terminals are connected The gateway modifies the address of the opposite end of the calling side and the called side of the media stream to the address of the edge relay gateway.

In the above solution, step 1) further includes: the edge relay gateway searches its own user information database according to the called user number carried in the received call access request, and determines whether the called user number exists, If it exists, the calling and called user terminals belong to the edge relay gateway; if not, the called user terminal does not belong to the edge relay gateway.

In the above solution, the calling user terminal and the called user terminal are connected to the same access gateway, or are connected to different access gateways.

In the two methods provided above, the access gateway may be an access media gateway, or an integrated access device, or a combination thereof.

The system of the present invention integrates the existing technology and the current status of the network of the new operator, and can better meet the needs of the operator, so that the new operator can use the IP network to comprehensively access a variety of services with only a small investment. At the same time, the implementation of the solution in the present invention mostly adopts more mature technology, and the newly added part only completes the functions of transparent transmission and conversion, so it has better stability and is easy to implement.

Because the gateway equipment used in the networking is the same as the equipment in the next generation network (NGN), then when the NGN matures, it only needs to introduce a new Media Gateway Control (MGC), and the existing networking can be It becomes the standard NGN network of MGC + IAD / AMG and MGC + TMG + LE. Therefore, operators can invest in one step and eventually transition to NGN smoothly, which is in line with the long-term interests of operators and the direction of technology development.

釆 After using the media stream processing method of the present invention, when users under the same edge relay gateway talk to each other, taking the calling and called users belonging to two different access media gateways as examples, the calling process is the calling user terminal 1> Caller accesses the media gateway 1> Called accesses the media gateway—) The called user terminal only needs to go through PCM ~> IP "~> PCM conversion, that is, only one encoding and decoding, reducing the gateway side Conversion, thereby improving performance parameters such as call quality and latency, reducing Following the gateway's processing burden, it can also save egress bandwidth, improve the carrying capacity of the IP access system for large traffic volumes, and also meet the special debugging and testing needs of operators or equipment manufacturers.

The invention can simultaneously support the realization of voice services, all existing data and multimedia services. Brief description of the drawings

Figure 1 is a schematic diagram of the composition of the IP access system of the present invention;

FIG. 2 is a schematic diagram of the IP access system of the present invention;

Figure 3 is an initial registration flowchart of IP access according to the present invention;

4 to 6 are schematic diagrams of call connection of a method for implementing a voice service according to the present invention;

FIG. 7 is a flowchart of the present invention.

FIG. 8 is a schematic diagram of a state in which media loop processing is performed by using an internal loopback according to the present invention; FIG.

FIG. 9 is a schematic diagram of a state after loopback in the present invention;

FIG. 10 is a flowchart of loopback processing in a preferred embodiment of the present invention. Mode of Carrying Out the Invention

The present invention will be described in further detail below with reference to the drawings and specific embodiments.

As shown in FIG. 1, the IP access system of the present invention is mainly composed of devices such as a local exchange (LE), an edge relay gateway (ETG), an integrated access device, and an access media gateway (AMG). Among them, the local switch is responsible for implementing call services and resource control; the edge relay gateway implements signaling conversion and media stream conversion functions; integrated access equipment and access media gateways are respectively used to connect narrowband side telephone users and broadband networks to achieve IP access. Here, the integrated access device and the access media gateway may be collectively referred to as an access gateway. Narrowband phone users are connected to the access gateways through subscriber lines, and each access gateway and edge relay gateway are connected to the IP metropolitan area network through Category 5 or network cables; the edge relay gateway is connected to the local switch through E1 / T1 trunk . The local switch controls the call of the telephone user under the jurisdiction of the access gateway through the edge relay gateway to implement voice services. Here, the media gateway control protocol (MGCP), the H.248 protocol, H.323 protocol or Start Session Protocol (SIP) communication; the local switch and the edge relay gateway use V5 signaling protocol, No. 7 signaling protocol, No. 1 signaling or Start Session Protocol (SIP) communication.

In fact, the IP access system of the present invention is based on the existing IP phone gateway functions, and adds MGCP / H.248 network side and V5 access side functions, so that the existing IP phone gateways are transformed into the edge. Following the gateway, the signaling conversion and media stream conversion functions are implemented.

In terms of hardware, since the media gateway / integrated access device, IP phone gateway, and switch already exist, the signaling interaction on the side of the media gateway / integrated access device uses the MGCP / H.248 protocol, and the switch side The signaling interaction uses the V5 / N0.7 protocol, so the implementation core of the present invention is based on the existing IP telephone gateway, and the conversion between the MGCP / H.248 protocol and the V5 / No.7 protocol is implemented by software. Complete the connection between the two. In this way, the existing IP telephony gateway is also transformed into an edge relay gateway. After interacting through signaling, the edge relay gateway creates an RTP channel connected to the access media gateway / integrated access gateway to carry the user's voice.

As shown in FIG. 2, the IP access system of the present invention is characterized in that: on the local switch side, the edge relay gateway and the access media gateway / integrated access device (ETG + AMG / IAD) behave as an access network (AN ); On the side of the access media gateway / integrated access device (AMG / IAD), the local switch and the edge relay gateway (LE + ETG) behave as a media gateway controller (MGC). At this time, the edge relay gateway is responsible for completing the corresponding protocols, such as the conversion between MGCP H.248 signaling and V5 signaling, and replaces the original V5 access method with IP access at the access layer. Then, based on the original software, the MGCP / H.248 protocol stack, the V5 layer 2 and layer 3 protocols need to be transplanted, and the corresponding protocol stack adaptation and conversion control functions need to be added.

For IP access, it includes: the initial registration process and the call connection process.

In the present invention, an access gateway is used as an access media gateway as an example. The initial registration process of IP access is shown in FIG. 3. After the relevant data is configured on the edge relay gateway, the access media gateway is powered on and the initialization is started. registration process:

After the user terminal initiates the initial registration process to the access media gateway where it is located, the access media The gateway sends a normal status endpoint status report message RSIP to the edge relay gateway; the edge relay gateway returns a response message RSIP ACK after receiving the RSIP message, and the edge relay gateway sends an endpoint audit message AUDIT to the access media gateway, auditing each Endpoint users; When the access media gateway completes the audit, it returns an audit response AUDIT ACK to the edge relay gateway, and the response is normal; the edge relay gateway reports the ASL user channel normal message ASL CHANNEL REPORT OK to the local switch, and simultaneously reports to the access The media gateway sends an RQNT (Off hook) message to detect the off-hook signal of the user, and waits for the call to be initiated. After the initialization registration process, the user completes the state initialization and registration on the switch and the edge relay gateway, and becomes a telephone user who can make normal calls.

Still taking the access gateway as the access media gateway as an example, the call connection process of IP access in the present invention is shown in FIG. 7, and referring to FIG. 4 to FIG. 6, where FIG. Schematic diagram of the connection process of the called user terminal 2. Fig. 5 'is a schematic diagram of the connection establishment between the called user terminal 2 and the local exchange. Fig. 6 is a schematic diagram of the connection setup for the called party. The solid line with arrows in the figure indicates signaling. The dotted line indicates the RTP channel. The call connection process includes the following steps:

Step 701: The calling user side, that is, the access media gateway 1 of the user terminal 1, reports the call event of the calling user to the edge relay gateway.

When the caller initiates a call, the access media gateway detects the off-hook signal of the calling user and reports the off-hook event to the edge relay gateway. Here, the format of the call event message reported by the access media gateway is Media Gateway Control Protocol (MGCP). ) Or H.248 protocol format.

Step 702: The control function module of the edge relay gateway establishes a real-time transmission protocol (RTP) channel between the edge relay gateway and the access media gateway, and converts the received call event protocol format to a local switch.

After the edge relay gateway receives the off-hook event report, the control function module of the edge relay gateway completes the establishment of the RTP channel and the resource allocation between the access media gateway and the edge relay gateway, and converts the message format of the user off-hook event. Report to local switch for V5 message format. This way, in Before the user dials, an RTP channel has been established between the access media gateway and the edge relay gateway to carry the voice channel.

Step 703: The control function module of the edge relay gateway on the calling user side uploads the user address to the local switch.

After the calling user dials, the edge relay gateway uploads the user's phone number to the local switch through signaling or an established RTP channel, and the local switch connects according to the called user's number, and at the same time, a ringback tone is returned to the caller. When passing through the edge relay gateway, it is converted into IP packets by Pulse Code Modulation (PCM), transmitted to the access media gateway through the RTP channel, and then converted into PCM signals by the access media gateway and put into the calling user to listen.

Step 704: The local switch delivers an instruction about the call event to the edge relay gateway on the user side according to the called user address. In this embodiment, the local switch issues an instruction to play a ring tone.

Step 705: The control function module of the called user-side edge relay gateway converts the indicated protocol format, and sends it to the called user-side access media gateway, and sends the edge relay gateway and the access medium to the access media gateway. A real-time transmission protocol (RTP) channel is established between the gateways.

When the edge relay gateway on the called side receives an instruction from the local switch to release ringing tones, it converts the V5 protocol format to the MGCP / H248 protocol format, and notifies the called user to access the media gateway to release the ringing to the user. An RTP channel between the called side edge relay gateway and the access media gateway is also established. In this way, before the called user goes off-hook, the RTP channel between the called side edge relay gateway and the access media gateway has also been set up.

Step 706: The access media gateway on the called user side notifies the called user of the call event. The access media gateway on the called user side plays a ring tone to the called user, and the call connection is completed.

Step 707: The calling user and the called user carry a voice service stream through an RTP channel. When the called user answers the call, the calling party and the called party enter the call state. The user's voice is converted through the media stream of PCM->IP->PCM->IP-> PCM. The RTP channel between the access media gateways carries the transmission, as shown in Figure 6.

Wherein, the control function module of the edge relay gateway described in step 702 establishes a real-time transmission protocol (RTP) channel between the edge relay gateway and the access media gateway, which may be performed in any step between steps 701 and 705. carry out.

When one party hangs up, the RTP channel corresponding to that party is closed, and related resources are released. At the same time, the busy tone is released to the unhanged user through the local switch. After the user hangs up, the corresponding RTP channel of the party is also closed and the related resources are released, and the call flow is ended.

It can be seen from the above implementation process that user terminal 1-> access to media gateway 1 → edge relay gateway → local switch → edge relay gateway "calling media gateway 2 to user terminal 2", the corresponding process The media stream of a call needs to be converted from PCM → IP → PCM → IP ~ PCM. This means that during the call, two PCM ^ IP codec operations are required, and the codec operation will affect the voice Quality and latency.

In order to improve the voice quality and reduce the delay, for the case where the calling party and the called party are under the same edge relay gateway, the present invention also proposes an internal loopback method to solve the problem of two codecs in the IP access technology. Specifically, when a calling user calls a telephone terminal of another called user through a telephone terminal, as shown in FIG. 10, after receiving the call access request initiated by the calling user, the edge relay gateway Analysis of the called user number, that is, searching the called user number in the user information database of the edge relay gateway; if the called user number is not found in the database, it means that the called user number does not belong to the edge relay gateway, At this time, follow-up operations are performed according to the obtained normal RTP address of the media stream receiver, and there is no need to modify or reset the address of the opposite end of the media stream master and callee. If the called user number is found in the database, it indicates that the master If the called user number belongs to the same edge relay gateway, the peer address of the media stream is set or modified.

When specifically setting or modifying, first determine whether the access media gateways of the calling and called sides have obtained the RTP address of the media stream receiver. If the RTP address of the receiver of a certain stream is obtained, the edge relay gateway instructs the access media gateway on the calling and called sides, and the corresponding access media gateway calls the media stream. The peer address on the side is changed to the IP address and port number corresponding to the called user, and the peer address on the called side of the media stream is changed to the IP address and port number corresponding to the calling user. Media streams are directly connected; if the RTP address of the media stream receiver is not obtained, the edge relay gateway instructs the calling and called sides to access the media gateway, and the corresponding access media gateway sends the corresponding address of the media stream to the calling side. Set the IP address and port number corresponding to the called user, and set the peer address on the called side of the media stream to the IP address and port number corresponding to the calling user, so that the media stream between the calling and called parties is directly connected .

Figures 8 and 9 show the states after internal loopback and when the internal loopback is suspended. The lines with arrows in each figure indicate the flow of media streams. The dashed lines in Figure 8 indicate the edge relay gateway and the access media gateway. The thick solid line indicates the newly created media flow channel between the access media gateways. The thick dashed line in FIG. 9 indicates the interrupted media flow channel between the access media gateways.

As shown in FIG. 8, through the above-mentioned setting or modification of the address of the calling party and the calling party, the calling channel of the called party can be directly connected, and the internal loopback of the IP network is completed. After the internal loopback of the IP network, the voice channel is directly transmitted from the media stream channel between the two access media gateways, that is, the access media gateway 1 and the access media gateway 2, and between the media gateway and the edge relay gateway. The path is no longer processing voice, but only reserved for switching.

As shown in FIG. 9, when the calling or called user hangs up / shoots or an abnormal situation occurs, the edge relay gateway controls the access media gateway where the calling and called users are located, that is, the connection where the calling user terminal 1 is located. Enter media gateway 1 and access media gateway 2 where the called user terminal 2 is located, and modify the address of the peer end of the calling side and the called side of the media stream to the address of the edge relay gateway, thereby interrupting the two access media gateway The speech channel between them returns to the normal state shown in FIG. 6. After the internal loopback is suspended, the call is still processed according to the original processing flow of IP access.

In order to realize the internal loopback of the IP network, it is necessary to establish the association between the calling party and the called party within the edge relay gateway. The edge relay gateway itself must know where a new call is to be called, and then according to the location of the called user terminal Decide whether to perform internal loopback. Currently in IP access, for edge Following the position of the gateway in the network, it is only a transmission component, not a switching component. After the voice channel is created, the number dialed by the user is sent to the local switch through transparent transmission, and the local switch will respond accordingly. Number analysis.

Therefore, if the internal loopback of the IP network is to be realized, the edge relay gateway itself must implement a simple number resolution function and have a control function of state transition from the general state to the internal loopback. ₅ required for this purpose take the following measures: 1) increasing the ETG properties: whether internal loopback; 2) to increase the allocation order, number length; 3) increase the loopback control function. In addition, all users under the edge relay gateway and their information, such as phone numbers, have been stored in its database, so it is only necessary to add a query interface.

After the edge relay gateway is configured to support the internal loopback mode, when the user dials the number, the edge relay gateway reports the number to the local switch and also reports it to the loopback control module. The loopback control module detects whether the number is currently waiting for receiving the number. If it is in the state of waiting for receiving the number, it collects the number. After receiving a predetermined number of digits, the state is set to the state of waiting for the loopback. Query the database. If this number is found, it indicates that the call belongs to the same edge relay gateway.

The loopback control module is quite an internal call control unit. When it is found that the calling and called users are all users of the same edge relay gateway, it orders to loop the media stream on the IP side to reduce one encoding and decoding. . The command to make the media stream loop is sent to the calling and called parties respectively. After receiving the command, the corresponding control module will cut off the RTP channel between the edge relay gateway and the access media gateway, and no longer send and receive RTP packets. Operation, of course, you can still restore the original state when you receive the restore command. At the same time, the edge relay gateway controls the access to the media gateway to modify the peer address of the media stream, change the peer address on the calling side to the IP address and port number corresponding to the called user, and change the peer address on the called side to The corresponding IP address and port number of the calling user, so that the media streams between the calling and called parties are directly connected, and the internal loopback of the network is completed.

The internal loopback of the present invention is not visible to the switch, and there is no change with the control channel of the switch. When a user hangs up, a fork event, or any abnormality occurs, the loopback control mode The block switches the control media stream back to the state before the modification.

The above embodiments have described the case where internal loopback is performed on calls between user phone terminals under the same edge relay gateway and under two different access media gateways. ₅ In fact, the solution of the present invention is not It is limited to the above-mentioned embodiment. For example, the location of the calling user terminal and the called user terminal may be in the same access media gateway ₅ or in the same integrated access device, or in two different integrated access devices, or a user is connected. To the media gateway and another user to the integrated access device.

In short, the above description is only the preferred embodiments of the present invention and is not intended to limit the present invention. Any modification, equivalent replacement, and improvement made within the spirit and principle of the present invention shall be included in the present invention. Within the scope of the invention.

Claims

Claim

1. An IP access system, including an IP network, characterized in that the system further includes: a local switch, which is used to implement call services and control resources;

More than one access gateway is used to connect a broadband network and a user terminal to implement IP access; an edge relay gateway is provided in the IP network and used to implement signaling conversion and media stream conversion; the local switch passes a relay chain The channel is connected to the edge relay gateway and interacts through a narrowband signaling protocol; the access gateway is connected to the edge relay gateway through an IP network.

2. The IP access system according to claim 1, wherein the edge relay gateway further comprises a loopback control module that controls a media stream to perform a self-loop on the IP side.

3. The IP access system according to claim 1, wherein: between the access gateway and the edge relay gateway is a Media Gateway Control Protocol (MGCP) link, or a H.248 protocol link, or H. 323 protocol link, or Start Session Protocol (SIP) link.

4. The IP access system according to claim 1, wherein the narrowband signaling protocol comprises V5 signaling, or No. 7 signaling, or No. 1 signaling, or PRA signaling protocol.

5. The IP access system according to any one of claims 1 to 4, characterized in that: the access gateway is an access media gateway (AMG), or an integrated access device (IAD), or a combination thereof.

6. A method for implementing voice services through IP access, characterized in that the method includes the following steps:

C. The voice service flows between the calling user terminal and the called user terminal pass through Time Transmission Protocol channels carry transmissions.

7. The method according to claim 6, wherein the step B specifically comprises:

B2. The edge relay gateway on the calling user terminal side reports the called user number to the local switch, and the local switch sends an instruction about the call event to the edge relay gateway on the user side according to the called user number;

B3. After the edge relay gateway on the called user terminal converts the indicated protocol format, it is delivered to the access gateway on the called user terminal side, and between the edge relay gateway and the access gateway. Establishing a real-time transmission protocol channel (RTP) channel;

B4. The access gateway on the called user terminal side notifies the called user terminal of the call event.

8. The method according to claim 6 or 7, characterized in that: the edge relay gateway and the access gateway adopt a Media Gateway Control Protocol (MGCP), or an H.248 protocol, or an H.323 protocol or Initiation Session Protocol (SIP) format; the V5 signaling, or # Ί signaling, or # 1 signaling, or PRA signaling protocol format is used between the edge relay gateway and the local switch.

9. The method according to claim 6 or 7, characterized in that: the access gateway is an access media gateway (AMG), or an integrated access device (IAD), or a combination thereof.

10. The method according to claim 7, characterized in that: in step B2, the edge relay gateway reports the called user number converted by its own protocol to the local switch through signaling, or passes the media through the real-time transmission protocol ( The RTP channel reports the called user number to the local switch.

11. The method according to claim 7 or 10, wherein: the edge relay gateway reporting the called user number to the local switch in step B2 further comprises:

B21. The local switch sends a ringback tone to the calling user terminal. B22. The edge relay gateway converts the pulse code modulation (PCM) ringback tone into an IP packet and transmits it to the access gateway through a real-time transmission protocol (RTP) channel; or instructs the access gateway to send a ringback tone through a control protocol;

12. The method according to claim 6, further comprising: when either the calling user terminal or the called user terminal terminates the service, the RTP channel corresponding to the terminating service party is closed and the related resources are released.

13. The method according to claim 6, further comprising: each user completing state initialization and registration in the edge relay gateway and the local switch through an access gateway before accessing.

14. A method for processing a media stream in an IP network, wherein the edge relay gateway executes the following steps after receiving a call access request initiated by a calling user terminal:

15. The method according to claim 14, wherein the step 2) further comprises the following steps:

21) First determine whether the access gateway of the calling and called sides has obtained the RTP address of the media stream receiver; 22) If the RTP address of the media stream receiving side has been obtained, the edge relay gateway instructs the access gateway of the calling and called side By changing the peer address on the calling side of the media stream to the IP address and port number corresponding to the called user terminal, and changing the peer address on the called side of the media stream to correspond to the calling user terminal The IP address and port number directly connect the media stream between the caller and the callee, and end the current process; if the RTP address of the media stream receiver is not obtained, proceed to step 23);

23) The edge relay gateway instructs the access gateway on the calling and called sides, which sets the peer address on the calling side of the media stream to the IP address and port number corresponding to the called user terminal, and sets the called side of a media stream The peer address is set to the IP address and port number corresponding to the calling user terminal, and directly connects the media stream between the calling and called parties.

16. The method according to claim 15, wherein in step 22), the edge relay gateway firstly transmits the real-time transmission between the edge relay gateway and the access media gateway to which the calling and called user terminals are connected. The protocol channel is cut off, and then the access gateway to which the calling and called user terminals are connected is modified to modify the peer address on the calling and called sides of the media stream.

17. The method according to any one of claims 14 to 16, characterized in that after the media stream between the calling party and the called party is directly connected in the step 2), the method further comprises: determining whether the calling party is received Or the called user terminal's on-hook / shooting signal or abnormal signal, if received, the edge relay gateway controls the access gateway to which the calling and called user terminal is connected, and the media stream on the calling side and the called side The peer address is changed to the address of the edge relay gateway.

18. The method according to any one of claims 14 to 16, wherein the step 1) further comprises: the edge relay gateway ^ according to the called party carried in the received call access request. User number, searching its own user information database to determine whether the called user number exists, and if it exists, the calling and called user terminals both belong to the edge relay gateway; if not, the called user terminal does not belong to The edge relay gateway.

19. The method according to any one of claims 14 to 16, wherein the calling user terminal and the called user terminal are connected to the same access gateway, or are connected to different access gateways.

20. The method of claim any one of claims 14 to 16, characterized in that said access gateway ₃ Access Media Gateway, or integrated access devices, or combinations thereof.