KR101636981B1 - Method for providing Voice over Long Term Evolution service in private network environment and gateway - Google Patents

Abstract

The present invention relates to a method for providing a voice over long term evolution (VoLTE) service with a service quality guarantee in a private network environment and a gateway thereof. According to the present invention, a private gateway providing a high quality VoLTE service in a private network environment comprises: a call processing module checking information of outgoing and incoming terminals, and information about a bandwidth required for a media stream when a VoLTE-based call connection is requested from a private base station, and generating a quality guarantee rule based on the checked pieces of information; and a communication interface module forming an S1 bearer for the VoLTE-based call with the private base station, wherein a stream bearer, which guarantees a bandwidth required in the S1 bearer and receives a VoLTE call packet, is formed in the S1 bearer on the basis of the quality guarantee rule generated in the call processing module.

Description

[0001] The present invention relates to a VoLTE service providing method and a gateway for providing VoLTE service in a private network environment,

The present invention relates to a method of providing a VoLTE service, and more particularly, to a method and a gateway for providing a VoLTE service with guaranteed quality of service in a private network environment.

Nowadays, corporations or specific organizations provide communication services using private switches to perform calls quickly and cheaply. In addition, a service for performing VoIP (Voice over Internet Protocol) -based call through a private branch exchange has also been disclosed. The following patent documents disclose an apparatus and method for controlling a VoIP call in a private branch exchange.

Further, a company or an organization may provide a private base station and a private gateway, and provide a Voice over Long Term Evolution (VoLTE) service using the private base station and the private gateway. Also, the private gateway provides a local breakout function to directly transmit the traffic of the terminal to the Internet network or a designated destination without going through the communication service provider network.

In a private network environment providing such a local breakout function, a conventional private gateway does not form a dedicated bearer for guaranteeing bandwidth when VoLTE-based call traffic is formed, . That is, VoLTE-based call traffic formed in a conventional private gateway does not receive a constant bandwidth, and packet loss and packet delay frequently occur, thereby degrading the quality of the VoLTE service.

Korean Patent Publication No. 10-2004-0052107

SUMMARY OF THE INVENTION The present invention has been proposed in order to solve such conventional problems, and it is an object of the present invention to provide a VoLTE service providing method and a gateway for guaranteeing call quality in a private network environment.

Other objects and advantages of the present invention will become apparent from the following description, and it will be understood by those skilled in the art that the present invention is not limited thereto. It will also be readily apparent that the objects and advantages of the invention may be realized and attained by means of the instrumentalities and combinations particularly pointed out in the appended claims.

According to a first aspect of the present invention, there is provided a private gateway for providing a high-quality voice over long term evolution (VoLTE) service in a private network environment. When a call connection request based on VoLTE is requested from a private base station, A call processing module for confirming necessary bandwidth information for the information of the called terminal and the media stream, and generating a quality assurance rule based on the confirmed information; And an S1 scaler for VoLTE-based communication with the private base station, wherein a bandwidth required in the S1 scaler is guaranteed based on the quality guarantee rules generated by the call processing module, And a communication interface module for forming a communication interface module in the S1 layer.

Preferably, the call processing module generates the quality guarantee rule recording GBR (Guaranteed Bit Rate) and MBR (Maximum Bit Rate) of the video and GBR and MBR of the audio, and the communication interface module transmits the GBR And a video scaler to which the MBR is applied and an audio scaler to which the GBR and MBR of the audio are applied are formed in the stream viewer.

In addition, the call processing module checks the required bandwidth of the audio and the required bandwidth of the video required for the media stream between the calling terminal and the called terminal, and based on the required bandwidth of the audio and the required bandwidth of the video, The MBR of the audio can be determined and recorded in the quality assurance rule.

The call processing module can confirm the required bandwidth of the audio and the required bandwidth of the video based on the SDP (Session Description Protocol) proposal information received from the calling terminal and the SDP response information received from the called terminal.

The private gateway may further include a media relay module for allocating an IP address and a port necessary for relaying media between the calling terminal and the called terminal.

In addition, the call processing module may transmit a call-related message to the calling terminal or the called terminal using the IP address and port allocated by the media relay module.

According to a second aspect of the present invention, there is provided a method of providing a high-quality voice over long term evolution (VoLTE) service in a private network environment, the method comprising: receiving a VoLTE-based call connection request from a private base station; Confirming the necessary bandwidth information for the information of the calling terminal and the called terminal and the media stream; Generating a quality assurance rule based on the verified information; And forming an S1 envelope for a VoLTE-based call with the private base station, wherein forming the S1 envelope comprises: guaranteeing bandwidth within the S1 envelope based on the quality assurance rule; Is formed in the S1 sheet-type winder.

The present invention has the advantage of improving the quality of the VoLTE service by forming a streamer with guaranteed maximum bandwidth and minimum bandwidth in the S1 layer and transmitting and receiving VoLTE based call packets through the stream browser have.

In addition, the present invention applies the bandwidth in which the audio bandwidth and the video bandwidth are summed to the stream browser, and only the call traffic is allowed to pass through the stream browser, thereby minimizing the packet loss and delay occurring in the VoLTE call, Service is provided.

BRIEF DESCRIPTION OF THE DRAWINGS The accompanying drawings, which are incorporated in and form a part of the specification, illustrate exemplary embodiments of the invention and, together with the description, serve to explain the principles of the invention. And shall not be construed as limited to such matters.
1 is a diagram illustrating a configuration of a VoLTE service providing system according to an embodiment of the present invention.
2 is a diagram illustrating a configuration of a private gateway according to an embodiment of the present invention.
FIG. 3 is a flowchart illustrating a method for providing a VoLTE service by forming a guaranteed bandwidth streamer in an S1 gateway in a private gateway according to an embodiment of the present invention. Referring to FIG.
4 is a diagram illustrating a method of applying QoS to VoLTE-related packets in a private gateway according to an embodiment of the present invention.
5 is a view of a stream bed formed in an S1 air baffle in accordance with an embodiment of the present invention.

The foregoing and other objects, features and advantages of the present invention will become more apparent from the following detailed description of the present invention when taken in conjunction with the accompanying drawings, in which: There will be. In the following description, well-known functions or constructions are not described in detail since they would obscure the invention in unnecessary detail. Hereinafter, a preferred embodiment of the present invention will be described in detail with reference to the accompanying drawings.

1 is a diagram illustrating a configuration of a VoLTE service providing system according to an embodiment of the present invention.

1, a VoLTE service providing system according to an embodiment of the present invention includes a mobile terminal 110, a private base station 120, a private gateway 200, a private branch exchange 130, an intranet server 140 and an extension terminal 150.

The mobile terminal 110 is capable of performing wireless communication with the private base station 120 and can perform voice call based on Voice over Long Term Evolution (VoLTE) with the extension terminal 150 connected to the private branch exchange 130. The mobile terminal 110 includes a tablet PC, a fourth generation mobile terminal, and the like, which is capable of processing a session initiation protocol (SIP) based message.

The extension station 150 is a terminal capable of forming a call session based on a SIP-based message, and is connected to the private branch exchange 130 to provide an extension call service to users.

The private branch exchange 130 is an apparatus capable of IP-based communication such as Internet Protocol Private Branch Exchange (IP-PBX), and is located in a specific business site and provides an extension call service to a user of the business site. In particular, when the PBX 130 receives an Invite message requesting a call connection to the extension station 150, the PBX 130 transmits an Invite message to the extension station 150 to form a call session. The private branch exchange 130 is connected to one or more extension terminals 150. The private branch exchange 130 stores an address (e.g., an IP address or a telephone number) of each extension terminal 150 and transmits a call related packet to the designated extension terminal 150 based on the address.

The intranet server 140 is a server that is located in a specific business location and provides information to employees and provides services related to business processes within the enterprise such as an e-mail service, an electronic bulletin board service, data provision for business processing, group scheduling service, .

The common LTE network 300 includes an MME (Mobile Mobility Entity), an S-GW (Serving GateWay), a P-GW (Packet Data Network GateWay) and an Internet network and performs a public Internet service and a call service.

The private base station 120 forms a radio link with the mobile terminal 110 and relays communication between the mobile terminal 110 and the private gateway 200. The base station 120 is a base station for establishing a wireless private network by forming a small cell. Also, the private base station 120 forms an S1 scaler with the private gateway 200, and transmits and receives a call related packet through the S1 scaler. Here, S1 is an interface between a base station and a gateway, and S1 is a bearer formed between a private base station 120 and a private gateway 200. As will be described later, the S1 envelope according to the present invention includes a stream envelope formed therein, and the stream envelope includes at least one of an audio envelope and a video envelope.

The private gateway 200 is a core device for constructing a private network, which restricts external access to the intranet server 140 and also performs a firewall function. The private gateway 200 is connected to the private base station 120 and transmits a packet received from the private base station 120 to the private branch exchange 130, the intranet server 140, or the common LTE network 300 . That is, when the packet is received from the private base station 120, the private gateway 200 checks whether the destination of the packet is a private network, and transmits the packet to the private branch exchange 130 or the intranet server 140 do.

In particular, the private gateway 200 forms a call session between the mobile terminal 110 and the extension terminal 150 when the mobile terminal 110 attempts to communicate with the extension station 150, A stream buffer is formed in an S1 scaler to guarantee quality of service, and the stream data is transmitted and received through the stream buffer. Here, the stream collector includes at least one of an audio buffer and a video buffer, which is an RTP dedicated buffer for transmitting a call related packet, that is, an RTP (Real Time Protocol) related packet in a VoLTE call. The audio player is a channel through which audio related packets are transmitted, and the video player is a channel through which video related packets are transmitted.

2 is a diagram illustrating a configuration of a private gateway according to an embodiment of the present invention.

2, the private gateway 200 according to an exemplary embodiment of the present invention includes a communication interface module 210, a storage module 220, a call processing module 230, and a media relay module 240 do.

The communication interface module 210 performs communication with the public LTE network 300 and the private base station 120, respectively. Upon receiving a packet from the private base station 120, the communication interface module 210 identifies the destination of the packet. If the packet is destined for the private network, the communication interface module 210 transmits the packet to the private branch exchange 120 or the intranet server 140, Lt; / RTI > In this case, if the packet is a call related packet, the communication interface module 210 delivers the packet to the call processing module 230, and if the destination of the packet is the intranet server 140, the communication interface module 210 transmits the packet to the intranet server 140.

In addition, the communication interface module 210 receives the quality assurance rules from the call processing module 230 and stores them in the storage module 220. In addition, the communication interface module 210 forms an S1 envelope with the private base station 120, and forms a stream envelope in the S1 envelope to which a constant bandwidth is applied based on the quality guarantee rules stored in the storage module 220. [ At this time, the communication interface module 210 confirms the Guaranteed Bit Rate (GBR) and the Maximum Bit Rate (MBR) of the audio and video recorded in the quality guarantee rule, and transmits the GBR and MBR of the audio to the audio scaler And the video archiver to which the GBR and MBR of the video are applied is included in the stream viewer. Accordingly, the GBR of the audio scaler and the GBR of the video scaler are applied to the stream viewer, and the MBR of the audio scaler and the MBR of the video scaler are applied to the stream viewer do. Here, Guaranteed Bit Rate (GBR) means a guaranteed minimum bandwidth, and Maximum Bit Rate (MBR) means a maximum available bandwidth.

The storage module 220 stores the IP address of the calling terminal, the audio port and video port of the calling terminal, the IP address of the called terminal, the audio port and video port of the called terminal, the GBR and MBR of the audio, and the GBR and MBR of the video Save the quality assurance rules.

The call processing module 230 functions to process call related packets received from the terminal. The call processing module 230 receives an Invite message, which is a call related message, from the communication interface module 210, and transmits the IP address and port information for receiving the media from the private branch exchange 130 to the media relay module 240, and transmits an Invite message in which the allocated IP address and port information are recorded to the private branch exchange 130, thereby forming a call session. Upon receiving the 200 OK message from the private switchboard 130, the call processing module 230 transmits the IP address and port information used when the media terminal 120 receives the media from the mobile terminal 110 to the media relay module 240, And transmits a 200 OK message in which the allocated IP address and port information are recorded to the private base station 120.

On the other hand, the call processing module 230 generates a quality assurance rule based on the information recorded in the Invite message and the 200 OK message, and transmits the quality assurance rule to the communication interface module 210. At this time, the call processing module 230 receives the IP address of the calling terminal, the audio port and the video port of the calling terminal, the IP address of the called terminal included in the 200 OK message, the audio port of the called terminal, Then, the MBR and GBR of the audio and the MBR and GBR of the video are mapped and recorded in the quality assurance rules.

Preferably, the call processing module 230 determines the MBR of the video and the MBR of the audio based on the required bandwidth of the audio and the required bandwidth of the video recorded in the SDP (Session Description Protocol) information of the Invite message and the 200 OK message Record in quality assurance rules. At this time, the call processing module 230 checks the bandwidth field (i.e., b = AS or b = TIAS) of the video and audio included in the Invite message and the 200 OK message, If b = AS ', a value obtained by multiplying the bandwidth recorded in each field by 1000 is recorded in the quality guarantee rule as the MBR of the video and the MBR of the audio. On the other hand, if the determined video bandwidth field and the audio bandwidth field are 'b = TIAS', the call processing module 230 records the bandwidth recorded in each field in the quality guarantee rule as the MBR of the video and the MBR of the audio.

In addition, the call processing module 230 records a value having a bandwidth equal to or less than the maximum bandwidth in the quality assurance rule with a minimum bandwidth. The call processing module 230 may record a predetermined value as an audio GBR and a video GBR in the quality guarantee rule and write the weighted bandwidth to each MBR in the quality guarantee rule as audio GBR and video GBR have.

The media relay module 240 allocates an IP address and a port necessary for media transmission and reception, and performs a function of relaying media generated during a call. That is, the media relay module 240 allocates an IP address and a port to be used when receiving the media from the private branch exchange 130, and transmits the IP address and the IP address used when the media is received from the mobile terminal 110 Allocate for port information. In addition, when a call is established between the calling terminal and the called terminal, the media relay module 240 relays the voice packet and the video packet transmitted and received between the calling terminal and the called terminal.

FIG. 3 is a flowchart illustrating a method for providing a VoLTE service by forming a guaranteed bandwidth streamer in an S1 gateway in a private gateway according to an embodiment of the present invention. Referring to FIG.

In the description with reference to FIG. 3, it is assumed that the calling terminal is the mobile terminal 110 and the called terminal is the extension terminal 150.

Referring to FIG. 3, the private base station 120 receives the Invite message from the mobile terminal 110 and transmits the Invite message to the private gateway 200 (S301). At this time, the private base station 120 forms a GTP tunnel with the private gateway 200, and transmits an Invite message to the private gateway 200 through the GTP tunnel. The Invite message includes an SDP (Session Description Protocol) proposal information for negotiating a multimedia session parameter in an Invite message, and the SDP proposal information includes an originating terminal , Codec information of the mobile terminal), audio port information and video port information, required bandwidth of audio, and necessary bandwidth of video.

Then, the communication interface module 210 of the private gateway 200 checks the destination address of the Invite message and confirms whether the destination address is a private network. If the destination address of the Invite message is not a private network, the private gateway 200 forwards the Invite message to the MME or the S-GW included in the public LTE network 300.

If the destination address of the Invite message is a private network, the communication interface module 210 transfers the Invite message to the call processing module 230 in step S303. The call processing module 230 receives the Invite message from the private branch exchange 130 media) to the media relay module 240 (S305). That is, the call processing module 230 requests the media relay module 240 to allocate the IP address and the port information used when communicating with the PABX.

Then, the media relay module 240 allocates the IP address, the audio port, and the video port used when receiving the media from the private branch exchange 130, and transmits the allocated IP address and port to the call processing module 230 (S307).

Next, the call processing module 230 transmits a new Invite message to the private branch exchange 130 to create a call with the private branch exchange 130 (S309). That is, the call processing module 230 generates a new Invite message and transmits the new Invite message to the PABX 130 to form a private network-based call session with the PABX 130. In step S307, the IP address, audio, and video port information allocated from the media relay module 240 are included in the Invite message including source information (i.e., source information) ). Then, the call processing module 230 receives the 200 OK message in response to the Invite message (S311). The 200 OK message includes SDP response information, and the SDP response information includes the audio port and the video port of the extension station 150, the required bandwidth of the audio of the called party, and the required bandwidth of the video.

Next, the call processing module 230 requests the media relay module 240 to allocate IP address and port information to be used when receiving the media from the mobile terminal 110 (S313). Then, the media relay module 240 allocates an IP address and a port to be used when receiving the media from the mobile terminal 110, and transmits the allocated IP address and port to the call processing module 230 (S315).

Then, the call processing module 230 generates a quality assurance rule based on the Invite message received from the private base station 120 and the 200 OK message received from the private branch exchange 130 (S317). The call processing module 230 transmits the IP address of the mobile terminal 110 included in the Invite message received from the private base station 120 to the audio port and the video port of the mobile terminal 110, The IP address of the called terminal, the audio port and the video port, the MBR and GBR of the audio, and the MBR and GBR of the video are mapped. At this time, the call processing module 230 determines the MBR of the video and the MBR of the audio based on the required bandwidth of the audio and the required bandwidth of the video, respectively, including the SDP proposal information of the Invite message and the SDP response information of the 200 OK message, Record in the guarantee rules. The call processing module 230 checks the bandwidth field (i.e., b = AS or b = TIAS) of the video and audio included in the Invite message and the 200 OK message, and if the video bandwidth field and the audio bandwidth field are 'b = AS ', a value obtained by multiplying the bandwidth recorded in each field by 1000 is recorded in the quality guarantee rule as the MBR of the video and the MBR of the audio. On the other hand, if the determined video bandwidth field and the audio bandwidth field are 'b = TIAS', the call processing module 230 records the bandwidth recorded in each field in the quality guarantee rule as the MBR of the video and the MBR of the audio.

In addition, the call processing module 230 may record the preset values as the audio GBR and the video GBR in the quality guarantee rule, and write the weighted bandwidth to each MBR in the quality guarantee rule as audio GBR and video GBR . Meanwhile, when the audio required bandwidth included in the SDP proposal information of the Invite message and the audio required bandwidth included in the SDP response information of the 200 OK message are different from each other, the call processing module 230 transmits the higher audio required bandwidth to the audio MBR Or the audio required bandwidth recorded in the SDP response information of 200 OK can be used as the MBR of the audio. Likewise, when the video required bandwidth included in the SDP proposal information of the Invite message and the video required bandwidth included in the SDP response information of the 200 OK message are different from each other, the call processing module 230 transmits a higher video required bandwidth or a 200 OK message The video required bandwidth recorded in the SDP response information can be used as the MBR of the video.

When the quality assurance rule is generated, the call processing module 230 transmits the generated rules to the communication interface module 210 (S319). The communication interface module 210 transmits the rules received from the call processing module 230 To the storage module 220 (S321).

Then, the call processing module 230 transmits the 200 OK message including the SDP response information to the private base station 120 using the communication interface module 210 (S323). At this time, the call processing module 230 generates SDP response information in which the IP address and port allocated from the media relay module 240 are recorded as the source IP address and port information, and transmits a 200 OK message including the SDP response information To the private base station (120).

Next, the private base station 120 transmits the 200 OK message to the mobile terminal 110, and upon receiving the ACK message from the mobile terminal 110, the private base station 120 transmits the message to the private gateway 200 (S325). The communication interface module 210 of the private gateway 120 transfers the ACK message to the call processing module 230 and the call processing module 230 transmits the ACK message to the extension terminal 150 via the private branch exchange 130 send.

When the ACK message is transmitted to the extension station 150, a call session is established between the mobile station 110 and the extension station 150.

The communication interface module 210 of the private gateway 200 can receive the call related packet from the mobile terminal 110. [ Then, the communication interface module 210 of the private gateway 200 checks the source address and port of the packet, the destination address and port, and the multimedia type (i.e., audio and video) of the packet, The MBR and GBR are confirmed in the quality assurance rules of the storage module 220. 3, the private gateway 200 receives both the audio related call packet and the video related call packet from the mobile terminal 110 and transmits the GBR and MBR of the audio and the GBR and MBR of the audio to the storage module 220, As shown in Fig.

Next, the communication interface module 210 forms an S1 envelope 30 with the private base station 120 as the audio related packet and the video related packet are received from the mobile terminal 110, A stream collector 31 and a general collector (not shown in the figure) comprising a collector 31a and a video collector 31b are formed in the S1 collector 30. At this time, the communication interface module 210 applies the GBR and MBR of the audio identified in the quality assurance rule to the audio scaler 31a, and transmits the GBR and MBR of the video identified in the quality assurance rule to the video scaler 31b . The GBR of the audio scaler 31a and the GBR of the video scaler 31b are applied to the stream scaler 31 and the MBR of the audio scaler 31a and the video scaler 31b are added to the stream buser 31. The MBRs of the stream buses 31,

When the S1 layer is formed in this manner, the media relay module 240 receives the audio packet of the mobile terminal 110 using the audio amplifier 31a with the guaranteed bandwidth as a path, and transmits the audio packet to the extension terminal 150 And transmits the video packet of the mobile terminal 110 to the extension terminal 150 using the required bandwidth of the video scaler 31b as a path. Similarly, the audio packet transmitted from the extension terminal 150 is transmitted to the mobile terminal 110 via the audio bus 31a, and the video packet transmitted from the extension terminal 150 is transmitted through the video scaler 31b And transmitted to the terminal 110. Packets that are not associated with the VoLTE service are also processed through the generic router in the S1 < RTI ID = 0.0 >

4 is a diagram illustrating a method of applying QoS to VoLTE-related packets in a private gateway according to an embodiment of the present invention.

5 is a view of a stream bed formed in an S1 air baffle in accordance with an embodiment of the present invention.

4 and 5, in a state where the quality assurance rule is stored in the storage module 220, the communication interface module 210 receives a call-related RTP packet from the media relay module 240 (S401). In the description with reference to FIGS. 4 and 5, it is described that the call-related RTP packet is an audio-related packet and a video-related packet.

Then, the communication interface module 210 checks the source address and port of the received packet, the destination address, the port and the multimedia type (i.e., audio and video) of the packet, and checks the MBR and GBR And the MBR and GBR of the video in the quality assurance rule of the storage module 220 (S403).

Next, the communication interface module 210 applies the MBR and GBR of the searched video to the video scaler 31b via the video-related RTP packet, and transmits the MBR and GBR of the searched video to the audio scaler 31a via the audio- The MBR and GBR of the searched audio are applied (S405).

Next, the communication interface module 210 forms a stream bus 31 in which the audio bus 31a and the video bus 31b are combined (S407), and the stream bus 31 is connected to the S1- (31) is applied to the S1 sheeter (30) so as to be included in the sheet bundle (30) (S409). The GBR of the audio scaler 31a and the GBR of the video scaler 31b are applied to the stream scaler 31 and the MBR of the audio scaler 31a and the video scaler 31b are added to the stream buser 31. The MBRs of the stream buses 31, The GBR and MBR of the audio bus 31a and the GBR and MBR of the video bus 31b are ensured in the stream bus 31 formed in the S1 sheet feeder 30.

Then, the communication interface module 210 transmits the audio and video-related RTP packets to the designated destination, that is, the calling terminal or the called terminal using the stream browser 31 (S411). That is, the communication interface module 210 receives the audio-related RTP packet transmitted from the mobile terminal 110 through the audio bus 31a and transmits the audio-related RTP packet to the media relay module 240, And receives the video-related RTP packet transmitted from the mobile terminal 110 and transmits the RTP packet to the media relay module 240. The communication interface module 210 receives the audio related RTP packet transmitted from the extension station 150 from the media relay module 240 and transmits the audio related RTP packet to the mobile terminal 110). The communication interface module 210 receives the video related RTP packet transmitted from the extension terminal 150 from the media relay module 240 and transmits the video related RTP packet to the mobile terminal 110). Preferably, the communication interface module 210 GTP encapsulates the packet when transmitting the packet through the stream browser 31 in the S1 envelope, and when receiving the packet through the stream browser 31, GTP reverse encapsulation.

Although the audio player and the video player are described as being formed at the same time in the above embodiment, the present invention is not limited to this, and one or more of the audio and video players, depending on the VoLTE call type, It can be formed in the substrate.

As described above, according to the present invention, a streamer with guaranteed maximum bandwidth and minimum bandwidth is formed in the S1 layer, and VoLTE-based call packets are transmitted and received through the stream browser, . In addition, the present invention applies the bandwidth in which the audio bandwidth and the video bandwidth are summed to the stream browser, and only the call traffic is allowed to pass through the stream browser, thereby minimizing the packet loss and delay occurring in the VoLTE call, Service.

While the specification contains many features, such features should not be construed as limiting the scope of the invention or the scope of the claims. In addition, the features described in the individual embodiments herein may be combined and implemented in a single embodiment. Conversely, various features described in the singular < Desc / Clms Page number 5 > embodiments herein may be implemented in various embodiments individually or in combination as appropriate.

Although the operations have been described in a particular order in the figures, it should be understood that such operations are performed in a particular order as shown, or that all described operations are performed to obtain a sequence of sequential orders, or a desired result . In certain circumstances, multitasking and parallel processing may be advantageous. It should also be understood that the division of various system components in the above embodiments does not require such distinction in all embodiments. The above-described program components and systems can generally be implemented as a single software product or as a package in multiple software products.

The method of the present invention as described above can be implemented by a program and stored in a computer-readable recording medium (CD-ROM, RAM, ROM, floppy disk, hard disk, magneto optical disk, etc.). Such a process can be easily carried out by those skilled in the art and will not be described in detail.

It will be apparent to those skilled in the art that various modifications and variations can be made in the present invention without departing from the spirit or scope of the invention. The present invention is not limited to the drawings.

110: mobile terminal 120: private base station
130: PABX 140: Intranet server
150: extension terminal 200: private gateway
210: communication interface module 220: storage module
230: call processing module 240: media relay module
300: Public LTE network

Claims (11)

As a private gateway that provides high-quality Voice over Long Term Evolution (VoLTE) services in a private network environment,
A call processing module for confirming the necessary bandwidth information for the information of the calling terminal and the called terminal and the media stream when the VoLTE based call connection is requested from the private base station and generating a quality guarantee rule based on the confirmed information; And
A stream browser for forming a S1 envelope for VoLTE-based communication with the private base station, wherein a bandwidth required in the S1 scaler is guaranteed based on the quality guarantee rules generated by the call processing module, and a VoLTE call packet is delivered; And a communication interface module for forming in the S1 layer a generic medium for delivering packets unrelated to the VoLTE call,
The call processing module generates the quality assurance rule in which GBR (Guaranteed Bit Rate) and MBR (Maximum Bit Rate) of video and GBR and MBR of audio are recorded,
Wherein the communication interface module forms a video scaler to which the GBR and the MBR of the video are applied and an audio scaler to which the GBR and the MBR of the audio are applied in the stream scaler and the GBR of the video and the GBR of the audio are added Applying the MBR of the video and the MBR of the audio to the stream viewer; and applying the MBR to the stream viewer. delete delete The method according to claim 1,
The call processing module,
Determining a required bandwidth of the audio and a necessary bandwidth of the video necessary for the media stream between the calling terminal and the called terminal and determining the MBR of the video and the MBR of the audio based on the required bandwidth of the audio and the required bandwidth of the video, Wherein the quality gateway records the information in the quality guarantee rule. 5. The method of claim 4,
The call processing module,
And the required bandwidth of the audio and the required bandwidth of the video are checked based on the SDP (Session Description Protocol) proposal information received from the calling terminal and the SDP response information received from the called terminal. The method according to claim 1,
And a media relay module for allocating an IP address and a port necessary for relaying media between the calling terminal and the called terminal,
Wherein the call processing module transmits a call related message to a calling terminal or a called terminal using an IP address and a port assigned by the media relay module. A method for providing a high quality Voice over Long Term Evolution (VoLTE) service in a private network environment,
Receiving a VoLTE-based call connection request from a private base station;
Confirming the necessary bandwidth information for the information of the calling terminal and the called terminal and the media stream;
Generating a quality assurance rule based on the verified information; And
And forming an S1 envelope for the VoLTE-based call with the private base station,
The generating of the quality assurance rules may include generating the quality assurance rules recording GBR (Guaranteed Bit Rate) and MBR (Maximum Bit Rate) of audio and GBR and MBR of audio,
The step of forming the S1 scaler comprises: providing a quality assurance rule based on the quality assurance rule, wherein a bandwidth is guaranteed in the S1 scaler and a stream scaler to which a VoLTE call packet is delivered and a general scaler to deliver a packet unrelated to the VoLTE call Wherein the GBR of the video and the GBR of the audio are formed in the stream browser, wherein the GBR of the video and the GBR of the audio are formed in the stream browser, Applying the summed GBR to the stream viewer, and applying an MBR in which the MBR of the video and the MBR of the audio are summed to the stream viewer. delete delete 8. The method of claim 7,
Wherein generating the quality assurance rules comprises:
The required bandwidth of the audio and the required bandwidth of the video necessary for delivering the media stream between the calling terminal and the called terminal are checked and the MBR of the video and the MBR of the audio are determined based on the required bandwidth of the audio and the required bandwidth of the video And recording the quality guarantee rule in the quality guarantee rule. 11. The method of claim 10,
Wherein generating the quality assurance rules comprises:
And the required bandwidth of the audio and the required bandwidth of the video are checked based on the SDP (Session Description Protocol) proposal information received from the calling terminal and the SDP response information received from the called terminal.

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