KR20120139302A - System for analyizing voip quality using packet forwarding type and method using the same - Google Patents

System for analyizing voip quality using packet forwarding type and method using the same Download PDF

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Publication number
KR20120139302A
KR20120139302A KR1020110059034A KR20110059034A KR20120139302A KR 20120139302 A KR20120139302 A KR 20120139302A KR 1020110059034 A KR1020110059034 A KR 1020110059034A KR 20110059034 A KR20110059034 A KR 20110059034A KR 20120139302 A KR20120139302 A KR 20120139302A
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KR
South Korea
Prior art keywords
packet
analysis server
quality analysis
quality
voip
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Application number
KR1020110059034A
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Korean (ko)
Inventor
유명환
김인천
김석
민천복
신성욱
Original Assignee
주식회사 케이티
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Priority to KR1020110059034A priority Critical patent/KR20120139302A/en
Publication of KR20120139302A publication Critical patent/KR20120139302A/en

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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L43/00Arrangements for monitoring or testing data switching networks
    • H04L43/08Monitoring or testing based on specific metrics, e.g. QoS, energy consumption or environmental parameters
    • H04L43/0823Errors, e.g. transmission errors
    • H04L43/0829Packet loss
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L43/00Arrangements for monitoring or testing data switching networks
    • H04L43/08Monitoring or testing based on specific metrics, e.g. QoS, energy consumption or environmental parameters
    • H04L43/0852Delays
    • H04L43/087Jitter
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04LTRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L43/00Arrangements for monitoring or testing data switching networks
    • H04L43/12Network monitoring probes

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  • Engineering & Computer Science (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Environmental & Geological Engineering (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)

Abstract

PURPOSE: A VoIP(voice over internet protocol) quality measurement system using a packet transmission method and method thereof are provided to directly transmit data packets which are transmitted and received between a VoIP terminal and an opponent terminal to an analysis server. CONSTITUTION: A quality analysis server(200) analyzes communication service quality using a packet. A user terminal transmits the packet transmitted and received in call connection to the quality analysis server. The user terminal includes a control unit, a packet creation unit, and a packet transmission unit. The control unit controls a call connection with an opponent terminal. The packet creation unit creates packets for connecting a call to the opponent terminal. The packet transmission unit transmits the packet created by the packet creation unit to the quality analysis server or the opponent terminal. [Reference numerals] (200) Quality analysis server; (A1,A2) Network; (B1,B2) RTCP FW

Description

System for analyizing VoIP quality using packet forwarding type and method using the same}

The present invention relates to a VoIP quality measurement system using a packet delivery method and a method thereof, and more particularly, to a VoIP terminal directly forwarding a packet to a packet quality analysis server.

Due to the complex network configuration and traffic characteristics, the Internet has been proposed and used in various ways.

The measurement methods proposed in the existing Internet networks are representative of active measurement methods and passive measurement methods.

Active measurement is a method of generating test packets or applications and observing the performance of the network. After adding test packets to the network, they measure the characteristics such as delay and loss that occur after the packets pass through the network. Such an active measurement method may deteriorate the network performance by generating additional traffic to the network, and the measurement performance may appear to be superior to or lower than that of the actual network due to the dynamic routing characteristics of the Internet network.

Manual measurement method calculates packet delay and packet loss, which are the traffic performance parameters of internet network, by generating a flow in the analysis system that analyzes network performance after collecting traffic flowing through the network using a packet collecting device without additional packet generation. That's the way.

However, the manual measurement method up to now requires a separate packet collecting device and a separate filtering device for filtering only the RTCP packets, and additionally constructing a communication line connecting the packet collecting device and the filtering devices. And there was a problem that a lot of maintenance costs are consumed.

An object of the present invention is to allow a VoIP terminal to directly transmit a data packet exchanged with a counterpart terminal to a quality analysis server.

Another object of the present invention is to selectively transmit a packet to a quality analysis server only when a VoIP terminal satisfies a specific condition.

VoIP quality measurement system using a packet delivery method according to the present invention for achieving the above object, the quality analysis server for analyzing the communication service quality using the packet, the packet transmitted and received when the call connection with the other terminal; It includes a user terminal to deliver to the quality analysis server.

In addition, the VoIP quality measurement method using the packet delivery method of the present invention, the process of the user terminal to connect the call with the other terminal, the process of generating a packet for the call connection to the other terminal and the quality analysis server, respectively And analyzing, by the quality analysis server, a communication service quality using the received packet.

As described above, the present invention provides high maintenance convenience and reduces cost consumption by allowing the packet to be analyzed for quality without separately providing a packet collecting device and a filtering device for collecting packets for VoIP quality analysis. It has an effect.

1 is a block diagram of a VoIP quality measurement system using a packet delivery method according to an embodiment of the present invention.
2 is a detailed configuration diagram of the VoIP terminal of FIG.
3A is a structural diagram of an RTCP packet transmitted to a counterpart terminal according to an embodiment of the present invention.
3b is a structural diagram of an RTCP packet forwarded to a quality analysis server according to an embodiment of the present invention;
4 is a flowchart illustrating a method for measuring VoIP quality using a packet delivery method according to an embodiment of the present invention.
5 is a flowchart illustrating a VoIP quality measurement method using a packet delivery method according to another embodiment of the present invention.

Hereinafter, a VoIP quality measurement system using a packet delivery method and a method thereof according to the present invention will be described in detail with reference to FIGS. 1 to 5.

 1 is a block diagram of a VoIP quality measurement system using a packet delivery method according to an embodiment of the present invention.

The VoIP terminal 101 transmits the RTCP packets to and from the quality analysis server 200 when communicating with the receiving VoIP terminal 102.

For this purpose, the VoIP terminals 101 and 102 have the detailed configuration as shown in FIG.

The VoIP terminals 101 and 102 transmit the RTCP packet to the quality analysis server 200 when a call is connected with the other terminal. To this end, the VoIP terminals 101 and 102 include a controller 110, a packet generator 120, a packet transmitter 130, and a forwarding condition determiner 140.

The controller 110 controls an operation for call connection. The controller 110 determines whether the VoIP terminal is in a hook-on state while in a call state, and in the case of a hook-off state, determines whether the hook-off state is maintained for at least 5 seconds, and when the hook-off state is maintained for at least 5 seconds, the RTCP packet forwarding in It is determined whether or not the set status is enabled. At this time, the controller 110 is previously set whether or not the RTCP packet forwarding when the VoIP terminal is initially installed.

The packet generator 120 generates an RTCP packet for performing a call when a call is connected. At this time, the RTCP packet transmitted to the counterpart terminal to perform a call has a structure as shown in FIG. 3A, and the RTCP packet for forwarding to the quality analysis server 200 has a structure as shown in FIG. 3B.

That is, as shown in Figure 3a, the RTCP packet to be delivered to the other terminal stores its own IP and the other terminal's IP in the IP field, the UDP field is an open port number for RTCP reception of the other terminal The data field stores the number of packet loss and the jitter value.

Meanwhile, as shown in FIG. 3B, the RTCP packet delivered to the quality analysis server 200 stores its own IP and IP information of the quality analysis server 200 in the IP field, and the quality analysis server 200 in the UDP field. The server port number is stored, the IP address of the other party's phone is stored in the DA field, and the cumulative packet count information and jitter information are stored in the Data field.

The packet transmitter 130 transmits the packet generated or combined by the packet generator 120 to a destination server or a terminal corresponding to the destination IP address.

The forwarding condition determination unit 140 determines that the quality is deteriorated when the number of packet loss stored in the data field of the RTCP packet is equal to or greater than the set value or the jitter is greater than or equal to the set value, and causes the controller 110 to determine the packet as a quality analysis server ( 200).

In this case, the number of losses refers to the number of losses of data packets transmitted and received during a call, and the higher the number of losses, the worse the quality of service. Jitter (jitter) is a deviation of the packet transmission delay in the network section, and high jitter means that the voice is interrupted and the communication quality of service is poor.

Hereinafter, a method for measuring VoIP quality using a packet delivery method according to an embodiment of the present invention will be described in detail with reference to FIG. 4.

First, when installing the first VoIP terminal, the IP address of the quality analysis server 200 to which the RTCP packet is delivered is set in advance.

The controller 110 of the VoIP terminal 101 transmits an RTP for a call when the terminal is in a call (S101), and determines whether it is in a hook-on state (S102).

The control unit 110 of the VoIP terminal 101 terminates the call in the hook-on state, and determines whether the hook-off state has elapsed for a predetermined time (5 seconds) in the hook-off state (S103).

As a result of the determination in step S103, when 5 seconds have elapsed, the controller 110 of the VoIP terminal 101 determines whether the RTCP packet forwarding is enabled (S104). ) Changes and recomposes the destination address of the RTCP packet to be sent to the other party in the call to the IP address of the quality analysis server 200, and the packet delivery unit 130 delivers the recombined packet to the quality analysis server 200 ( S105). The RTCP packet is transmitted to the counterpart terminal IP address (S106).

As such, when the present invention is set to the RTCP packet forwarding enabled state, the VoIP terminal 101 delivers the RTCP packet generated during the call with the counterpart terminal 102 to the quality analysis server 200 having the forwarding setting.

Therefore, the present invention sets the IP address of the quality analysis server 200 to which the RTCP packet will be delivered when the VoIP terminal is initially installed, and automatically reassembles the RTCP packet in the VoIP terminal during a call and delivers it to the quality analysis server 200. By doing so, it is not necessary to include a packet collecting device and a filtering device in the quality analysis server 200.

Hereinafter, a method for measuring VoIP quality using a packet delivery method according to another embodiment of the present invention will be described in detail with reference to FIG. 5.

First, the control unit 110 of the VoIP terminal 101 transmits the RTP for call connection (S101), and determines whether it is in a hook-on state (S202).

The controller 110 of the VoIP terminal 101 terminates the call if it is in the hook-on state, and determines whether the hook-off state has elapsed for a predetermined time (5 seconds) if not in the hook-on state (S203).

As a result of the determination in step S203, when 5 seconds have elapsed, the control unit 110 of the VoIP terminal 101 determines whether the RTCP packet forwarding enable state is set (S204). As a result of the determination, when the RTCP packet forwarding enable state is set, the forwarding condition determination unit 140 determines whether the accumulated packet loss count of the RTCP packet is increased by a specific number (5) or more compared to the previous RTCP packet (S205). When the specific number is increased by more than the previous RTCP packet, the packet forwarding unit 130 performs forwarding of the RTCP packet to the quality analysis server 200 (S207).

In addition, it is determined whether the jitter of the RTCP packet is increased to a specific value (5) or more (S206), and when the jitter is increased to a specific value or more, the packet forwarding unit 130 forwards the RTCP packet to the quality analysis server 200. It performs (S207).

Subsequently, the VoIP terminal 101 transmits the RTCP packet to the counterpart terminal 102 IP address (S208).

As described above, the present invention not only forwards the RTCP packet from the VoIP terminal to the quality analysis server 200 as shown in FIG. 4, but also determines the current packet state (loss number, jitter) of the VoIP terminal as shown in FIG. In this case, the TCP packet is forwarded to the quality analysis server 200. That is, by selectively forwarding the RTCP packet to prevent the traffic excessive to the quality analysis server 200.

101, 102: VoIP terminal
200: Quality Analysis Server
110:
120: packet generation unit
130: packet delivery unit
140: forwarding condition determination unit

Claims (11)

A quality analysis server for analyzing a communication service quality using a packet; And
User terminal that transmits the packet transmitted / received when the call is connected with the other terminal to the quality analysis server
VoIP quality measurement system using a packet delivery method comprising a.
The method of claim 1,
The user terminal,
A control unit controlling a call connection with the counterpart terminal;
A packet generation unit generating a packet for call connection with the counterpart terminal; And
Packet transmission unit for transmitting the packet generated by the packet generation unit to the quality analysis server or the counterpart terminal
VoIP quality measurement system using a packet delivery method characterized in that it comprises a.
The method of claim 2,
The packet generation unit,
In the case of a packet transmitted to the quality analysis server,
Packet forwarding using the packet forwarding method characterized in that the packet and data field values transmitted to the counterpart terminal are the same and the destination address is changed to the IP address of the quality analysis server and the destination port number is changed to the port number of the quality analysis server. Quality measurement system.
The method of claim 2,
Forwarding condition determination unit for determining whether the cumulative loss count value or jitter value stored in the data field value of the packet is greater than or equal to the set value
VoIP quality measurement system using a packet delivery method characterized in that it further comprises.
The method of claim 4,
The control unit,
And transmitting the packet to the quality analysis server when the cumulative loss number value or jitter value is greater than or equal to the set value according to the result of the determination of the forwarding condition determination unit.
A control unit controlling a call connection with a counterpart terminal;
A packet generator for generating a packet for the call connection;
A packet delivery unit for transmitting the packet generated by the packet generation unit to the quality analysis server or the counterpart terminal;
Forwarding condition determination unit for determining whether the cumulative loss count value or jitter value stored in the data field value of the packet is greater than or equal to the set value
Including a user terminal having a,
The control unit, according to the determination result of the forwarding condition determination unit, when the cumulative loss number value or jitter value is more than the set value, delivers the packet to the quality analysis server, VoIP quality using the packet delivery method Measuring system.
The method according to claim 6,
The packet generation unit,
In the case of a packet transmitted to the quality analysis server,
Packet forwarding using the packet forwarding method characterized in that the packet and data field values transmitted to the counterpart terminal are the same and the destination address is changed to the IP address of the quality analysis server and the destination port number is changed to the port number of the quality analysis server. Quality measurement system.
Connecting the call with the counterpart terminal by the user terminal;
Generating a packet for the call connection and transmitting the packet to the counterpart terminal and the quality analysis server; And
A process of analyzing a communication service quality by using the received packet in the quality analysis server
VoIP quality measurement method using a packet delivery method comprising a.
The method according to claim 8,
The process of generating a packet for the call connection and transmitting to the counterpart terminal and the quality analysis server, respectively,
The packet transmitted to the counterpart terminal and the data field value are the same and the destination address is changed to the IP address of the quality analysis server and the destination port number is changed to the port number of the quality analysis server to generate a packet transmitted to the quality analysis server. VoIP quality measurement method using a packet delivery method characterized in that.
The method of claim 8,
The process of generating a packet for the call connection and transmitting to the counterpart terminal and the quality analysis server, respectively,
It is determined whether the jitter value or the cumulative loss number value of the packet is greater than or equal to the set value, and if the jitter value or the cumulative loss number value of the packet is greater than or equal to the set value, the packet is delivered to the quality analysis server. VoIP Quality Measurement Method
The method of claim 8,
The process of generating a packet for the call connection and transmitting to the counterpart terminal and the quality analysis server, respectively,
Registering the packet to the quality analysis server in advance and registering the packet; And
Checking whether the packet is delivered to the quality analysis server when the packet is generated for the call connection with the counterpart terminal, and transmitting the packet to the quality analysis server when the packet is delivered to the quality analysis server.
VoIP quality measurement method using a packet delivery method characterized in that it comprises a.
KR1020110059034A 2011-06-17 2011-06-17 System for analyizing voip quality using packet forwarding type and method using the same KR20120139302A (en)

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Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2014104614A1 (en) * 2012-12-28 2014-07-03 주식회사에어플러그 Method and device for performing ip address-based telephone conversation through plurality of heterogeneous communication networks
WO2015174628A1 (en) * 2014-05-12 2015-11-19 주식회사 네이블커뮤니케이션즈 Volte quality measuring system and volte quality measuring method utilizing terminal agent
KR20160101259A (en) * 2015-02-16 2016-08-25 주식회사 엘지유플러스 Communication apparatus and communication quality measurement server, and control method thereof
US10623465B2 (en) 2016-01-25 2020-04-14 Byung Jin Moon Method for predicting call quality and call quality prediction service apparatus for performing the same

Cited By (4)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO2014104614A1 (en) * 2012-12-28 2014-07-03 주식회사에어플러그 Method and device for performing ip address-based telephone conversation through plurality of heterogeneous communication networks
WO2015174628A1 (en) * 2014-05-12 2015-11-19 주식회사 네이블커뮤니케이션즈 Volte quality measuring system and volte quality measuring method utilizing terminal agent
KR20160101259A (en) * 2015-02-16 2016-08-25 주식회사 엘지유플러스 Communication apparatus and communication quality measurement server, and control method thereof
US10623465B2 (en) 2016-01-25 2020-04-14 Byung Jin Moon Method for predicting call quality and call quality prediction service apparatus for performing the same

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