KR101174027B1 - A media quality probe system for internet phone telecommunication on the encrypted voip network - Google Patents

Abstract

PURPOSE: A media quality probe system for an internet phone telecommunication on an encrypted VOIP network is provided to exactly measure voice telecommunication quality from both a caller side and a receiver side. CONSTITUTION: A plurality of measurement terminals(30) receives internet telephone numbers. The measurement terminals include an internet call function. The measurement terminals calculate communication quality data through the communication to the internet telephone number. A server(40) performs communication through the measurement terminals and an SNMP(Simple Network Management Protocol) protocol. The server collects quality data from the measurement terminals or transmits communication command to the measurement terminals.

Description

A Media Quality Probe System for Internet Phone Telecommunication on the Encrypted VoIP Network}

The present invention relates to a media quality measurement system in an encrypted VoIP network for measuring the communication quality of Internet telephony between an internal network and an external network where a firewall is installed.

In general, Voice over Internet Protocol (VoIP) service method uses voice subscriber network such as xDSL network and cable network that is established to provide Internet service and corporate subscriber network such as dedicated line. IP data packet that can be transmitted is transmitted.

The packetized voice data traffic is mixed with the existing general data communication traffic and transmitted to the desired destination host through the ISP backbone network while sharing the network bandwidth and interval. In addition, voice data traffic generated by the other host is also transmitted in the same manner.

However, since IP packetized voice data also shares network resources with existing Internet traffic, problems such as data loss, excessive transmission delay, out-of-order and irregular packet arrival may occur when general data traffic is congested or network quality is degraded. In particular, it has a vulnerability that excessive packet loss is likely to occur because RTP packets are transmitted in an unconnected UDP protocol rather than a connected TCP protocol. This can seriously affect the subjective voice quality felt by VoIP subscribers.

Therefore, VoIP using the Internet network is a service that requires more real-time, unlike data transmission that requires flawlessness to guarantee voice call quality, and the degradation of call quality is an important issue due to the limitation of the real-time of the Internet network. .

Institutions providing VoIP services need to continuously measure and analyze call quality in order to resolve user complaints caused by poor voice call quality and to ensure effective call quality for service users.

In particular, unlike the PSTN which uses a proprietary line, such VoIP uses a flexible line, which is likely to cause a large amount of packet loss and delay depending on the network traffic of the Internet. Therefore, QoS (Service of Quality) is higher than that of the PSTN. As a result, there is a strong demand for accurate service quality measurement.

For the above reasons, examples of techniques for measuring the quality of a VoIP voice call are described in Korean Patent No. 10-0984844 (published on Oct. 4, 2010), "ＶFIIP terminal quality measuring system and method" (hereinafter, referred to as Prior Art 1). , [Registered Korean Patent No. 10-0967892 (published July 6, 2010), "Method for measuring the quality of Internet telephony"] (hereinafter, prior art 2), [Korea Patent Publication No. 10-2007-0060861 (Jul. 13, 2007). Publicly disclosed), " Method for measuring interactive interactive call quality in VoIP network " (hereinafter, referred to as prior art 3).

The prior art 1 is a technology for measuring the voice call quality by receiving a packet of the VoIP terminal by connecting a terminal measuring device to the network connected to the VoIP terminal, the prior art 2 is a measurement equipment connected to the Internet telephone It is a technique for measuring voice by transmitting voice data to the Internet telephone and measuring the data returned to the loopback. However, since the prior arts 1 and 2 both measure the voice call quality in one direction, there is a problem in that it is impossible to measure the delay or jitter caused when the sender and the receiver directly make the voice call.

In addition, the prior art 3 integrates E-Model and Perceptual Evaluation of Speech Quality (PESQ), an objective quality measurement method, for measuring end-to-end interactive call quality in a VoIP network. It is a technology that applies the E-Model of the method. However, since the prior art 3 proposes a new model for measuring call quality under the premise that both the transmitter and the receiver can measure call quality, there is a problem that the measurement terminal must be implemented with the new model as described above.

Therefore, there is a need for a technique for measuring call quality in end-to-end directions while using the conventional VoIP communication protocol.

In particular, in recent years, many companies, such as using the Internet established in the organization to configure the VoIP network. At this time, each company installs a firewall in the internal network for security and regulates the communication by encrypting the voice call in the VoIP network.

Therefore, a technique for more accurately measuring voice call quality in a firewalled and encrypted VoIP network in both directions is urgently needed.

SUMMARY OF THE INVENTION An object of the present invention is to solve the problems as described above, and to provide a media quality measurement system in an encrypted VoIP network for measuring the communication quality of the Internet phone encrypted and transmitted between the internal network and the external network where the firewall is installed. will be.

In order to achieve the above object, the present invention relates to a media quality measurement system in an encrypted VoIP network that measures the communication quality of Internet telephony between an internal network and an external network where a firewall is installed. As a measurement terminal, a plurality of measurement terminals are provided with an Internet telephone number and have an Internet telephone call function, and calculate communication quality data when making a call to the Internet telephone number. And a server that communicates with the measurement terminal through a Simple Network Management Protocol (SNMP) protocol and transmits a call command to the measurement terminal or collects quality data from the measurement terminal, wherein the protocol of the Internet telephony call is encrypted. And is transmitted.

In addition, the present invention provides a system for measuring a media quality in an encrypted VoIP network, wherein the server calls the measuring terminal (hereinafter referred to as a transmitting terminal) to a telephone number of another measuring terminal (hereinafter referred to as a receiving terminal) by scheduling. It is characterized by commanding a call.

The present invention provides a media quality measurement system in an encrypted VoIP network, the measurement terminal comprising: a call processing unit for forming a call path for an internet telephone call and transmitting and receiving a test voice; A quality measuring unit calculating quality data from the media including the test voice; A server communication unit communicating with the server through an SNMP protocol; And a control unit configured to receive a call command from the server to form a phone call or to transmit the quality data to the server.

In another aspect, the present invention provides a system for measuring media quality in an encrypted VoIP network, wherein the server communication unit transmits the quality data to the server or receives a call command from the server using an SNMP trap. It is characterized by.

In addition, the present invention is a media quality measurement system in an encrypted VoIP network, the quality measuring unit is characterized in that the quality of the media using the Real-time Transport Protocol (RTP).

In addition, the present invention is a media quality measurement system in an encrypted VoIP network, the quality measuring unit is characterized by measuring the quality of the media using the Real-time Transport Control Protocol (RTCP).

In addition, the present invention is a media quality measurement system in an encrypted VoIP network, the quality data is any one of R value, MOS, delay, jitter, packet loss rate, call success rate It is characterized by including the above.

As described above, according to the media quality measurement system in an encrypted VoIP network according to the present invention, an effect of more accurately measuring voice call quality in an encrypted VoIP network in which a firewall is installed is obtained in both directions.

1 is a diagram showing the configuration of an entire system for implementing the present invention.
2 illustrates a basic Internet telephony protocol in accordance with the present invention.
3 is a block diagram of a configuration of a measurement terminal for measuring media quality in an encrypted VoIP network according to an embodiment of the present invention.
4 is a flowchart illustrating a communication process between measurement terminals according to an embodiment of the present invention.
5 illustrates a message exchange order and message contents exchanged between a server and a measurement terminal according to an embodiment of the present invention.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS Hereinafter, the present invention will be described in detail with reference to the drawings.

In addition, in describing this invention, the same code | symbol is attached | subjected and the repeated description is abbreviate | omitted.

First, a configuration of a media quality measurement system in an encrypted VoIP network according to an embodiment of the present invention will be described with reference to FIG. 1.

As shown in FIG. 1, a media quality measurement system in a VoIP network according to an embodiment of the present invention includes a plurality of internal networks 21a and 21b, an internet network 22, and a server 40. Each of the internal networks 21a and 21b includes an IP switch 11, a router 12, and a firewall 13, and the internal networks 21a and 21b are connected to each other through an internet network 22 to communicate with each other. It is possible. In addition, the measurement terminal 30 is configured to be connected to each of the internal network (21a, 21b). In addition, the server 40 may further include a database 50 for storing necessary data such as quality data, analyzed data, and scheduling data.

Each internal network 21a, 21b is a network provided in one organization and has a firewall 13 to protect it from the outside. The firewall 13 may have a network address translation (NAT) function. In other words, public IP addresses are used externally through NAT, and private IP addresses are used internally.

The IP switch (IP PBX) 11 is a switch designed to use an external trunk line and an internal line as an IP line (LAN, WAN, etc.) in a solid line section of a conventional TDM scheme.

The measuring terminal 30 is provided with an Internet call function, and is given an IP address and an Internet telephone number and connected to another Internet telephone terminal or another measuring terminal with an Internet telephone call through the IP switch 11.

In addition, the measurement terminal 30 calculates communication quality data during an internet call with the other measurement terminal 30, and temporarily stores the calculated quality data. The measurement terminal 30 transmits the measured quality data to the server 40.

In this case, the measurement terminal 30 communicates with the server 40 through the SNMP (Simple Network Management Protocol) protocol. In addition, the measurement terminal 30 communicates with the server 40 by applying a hole punching (Hole Punching) technique to pass through the firewall (13).

In addition, the measurement terminal 30 encrypts and transmits data related to a voice call, and uses an encrypted VoIP protocol even when a call is connected. At this time, the SIP protocol is used for call connection, and the Real-time Transport Protocol (RTP) protocol is used for media connection for voice or video data transmission. In addition, Real-time Transport Control Protocol (RTCP) is used to maintain QoS of the RTP protocol.

As shown in Fig. 2, the SIP protocol is an HTTP-based protocol in which an Internet telephone terminal locates a location to identify a communication counterpart and specifies a procedure for creating, deleting, or modifying a media data communication session with each other. The protocol that encrypts this is the TLS security protocol.

In addition, RTP is a protocol for transmitting media (audio, video) to a counterpart in real time, and the protocol that encrypts it is sRTP. RTP (or sRTP) does not need to pass through an IP switch (IP-PBX switch) 11 unlike SIP because the RTP (or sRTP) has to be delivered to the other party through the shortest path through the IP network.

That is, the measurement terminal 30 performs encrypted communication by using a transport layer security (TLS) protocol to connect to a session initiation protocol (SIP). That is, the measurement terminal 30 sets the SIP protocol encrypted through the TLS protocol.

Through this, the measurement terminal 30 receives the session key (or symmetric key) and another measurement terminal to communicate with each other, and performs encrypted media communication.

On the other hand, the measurement terminal 30 calculates the quality data while performing encrypted VoIP communication. Quality data include R value, MOS, delay, jitter, packet loss, call success rate, and so on.

The server 40 transmits a call command to the measurement terminal 30 through the SNMP protocol or receives and collects quality data from the measurement terminal 30. In particular, the server 40 instructs the measurement terminal (hereinafter referred to as a transmission terminal) to make a telephone call to the telephone number of another measurement terminal (hereinafter referred to as a reception terminal) that is different from the measurement terminal.

The server 40 has a scheduling to give a call command to the measurement terminal 30 in advance, and transmits a call command to the measurement terminal 30 according to the scheduling. As described above, the measurement terminal 30 attempts a telephone call by a call command, and calculates quality data during call setup and during a call.

Therefore, call quality data is collected by call order scheduling. The server 40 collects quality data calculated from the measurement terminal 30. The collected quality data is used as data for measuring the quality of an internet telephone call. In particular, the server 40 collects quality data from both the transmitting side measurement terminal and the receiving side measurement terminal to more accurately measure the call quality in both directions.

Next, the configuration of the measurement terminal according to an embodiment of the present invention will be described in more detail with reference to FIG. 3.

As shown in FIG. 3, in one embodiment of the present invention, the measurement terminal 30 includes a call processing unit 31, a quality measuring unit 32, a server communication unit 33, and a control unit 34. In addition, it may be configured to further include a memory 35 for storing data.

The call processor 31 forms a call path for an internet telephone call, and transmits and receives a test voice.

The call processing unit 31 encrypts and receives data related to a voice call and uses an encrypted VoIP protocol even when a call is connected. As described above, the SIP protocol is used for call connection, and the RTP and RTCP protocols are used for media connection for voice or video data transmission.

In addition, the call processing unit 31 performs an encrypted communication using the TLS protocol for SIP connection. That is, the measurement terminal 30 sets the SIP protocol encrypted through the TLS protocol. Through this, the measurement terminal 30 receives the session key (or symmetric key) and another measurement terminal to communicate with each other, and performs encrypted media communication.

As shown in FIG. 4, messages transmitted / received between the measurement terminals 30 except for RTP media corresponding to voice or video in the SIP communication protocol are called control messages. In FIG. 4, it is indicated as TLS.

The call processor 31 performs an encryption process while performing SIP communication as shown in FIG. 5A. That is, the present invention is applied to perform the above encryption process while performing SIP communication, not the above encryption process. 5b illustrates the role of a message transmitted and received in steps of FIG. 5a.

The server of FIG. 5A is an IP exchanger (IP-PBX) 11 that is located between a terminal and a terminal and connects a calling terminal and a called terminal to VoIP through SIP communication. If the network becomes more complex between terminals, it can be connected through two or more servers.

In addition, the encryption method applied by the call processor 31 uses a public key-based RSA package, and can selectively apply AES and ARIA as an algorithm. To do this, it is configured by using the OpenSSL package in the SIP protocol stack and encrypted using AES or ARIA algorithm.

On the other hand, the call processor 31 processes media message encryption and sends and receives.

That is, as a method using sRTP in SIP-based VoIP communication, negogo is achieved through an SDP of 200 OK for INVITE. Also, at the bottom of the INVITE message, unlike other requests, "SDP" is attached.

Through this, the present invention can measure the quality of media while exchanging traffic information through encrypted communication.

Specifically, there are SIP Phones and H.323 Phones among the protocols for Internet telephony or VoIP communication. In recent years, SIP Phones are being used more and more. Here, the message of the phone using the SIP protocol is divided into a signaling part and a media part. The signaling part is called SIP, and when encrypted, it becomes TLS (Transport Layer Security).

In addition, when the RTP which is a media part is encrypted, it becomes sRTP. The encryption algorithms used in these two encryption processes are AES and ARIA. The earliest encryption in the Internet that actually implemented this algorithm was SSL, and this has evolved to become the TLS in use today. The package used for encryption is OpenSSL.

When using sRTP in SIP-based VoIP communication, Nego is established through SDES of INVITE, that is, 200 SDES of INVITE message. An example of this is as shown in FIG. 6. In FIG. 6, the m = audio and a = crypto parts are negotiated information.

On the other hand, the method used for encryption is as shown in FIG.

As encryption is added, an encryption encoding process is added at the source terminal, and an encryption decoding process is added at the receiving terminal. The present invention has also been extended (AES, ARIA) in the encryption method.

The quality measuring unit 32 calculates quality data from the media including the test voice. In particular, the quality measuring unit 32 measures the quality of the media using a real-time transport protocol (RTP) or a real-time transport control protocol (RTCP).

In this case, the calculated quality data includes R value, MOS, delay, jitter, packet loss, call success rate, and the like.

On the other hand, as previously seen, transmission and reception of media via SIP communication or RTP is encrypted and communicated. Therefore, the quality measuring unit 32 may measure the quality of the media while exchanging traffic information through encrypted communication.

In terms of quality measurement, since the quality is measured between the terminal and the terminal for the network end, it will be regarded as measuring the quality of the network located between each terminal as well as the terminal.

Strictly speaking, since it is RTP, that is, audio or image data, which is used as quality measurement data, quality measurement is performed on the path through which RTP flows, not through each server.

Analysis for quality measurement is performed on the RTP protocol data, and the RTP data is transmitted in small packet units. The packet sent from the transmitting terminal is received by the receiving terminal to calculate the quality measurement value.

The measured parameter values are then used to calculate the quality indicators by the ITU-T Recommendation E-Model.

Quality indicators include call success rate, R-Value, MOS, one-way delay, jitter, and packet loss rate.

The call success rate is the ratio of the normal calls that are successfully connected to the total attempted calls, and determines whether the call succeeds after comparing the status code values of the SIP Response packets with respect to the SIP INVITE packet.

R-Value is an end-to-end voice quality rating based on E-Model, a service performance indicator, and the default values and measured values (One-Way Delay, Packet Loss) for each of the 20 detailed parameters presented in the E-Model reference model. Etc.) The result value is calculated through application.

In addition, the one-way delay is a subjective evaluation value that scores the opinions of the subjects regarding the voice quality, and calculates the MOS value based on the R-Value to which the E-Model is applied.

Jitter is a change amount of the end-to-end unidirectional delay for the packet from the transmitting terminal to the receiving terminal, and is calculated by referring to the "timestamp" field of the RTP header and the "A" field of the SIP header.

Packet Loss Rate is the loss rate that occurs during RTP packet transmission after normal call connection of SIP. Packet loss referring to the "Sequence Number" field of RTP Header starting from a random value and increasing by 1 for transmitted RTP packet. Check.

That is, a quality indicator value obtained through the analysis of the RTCP SR block transmitted from the transmitting terminal is added to the RTCP RR block by the receiving terminal and transmitted. And if RTCP is not delivered, quality can be measured by RTP alone. RTCP SR / RR packet is transmitted at regular intervals during the test call termination or call.

The server communication unit 33 communicates with the server 40 in the SNMP protocol.

In addition, the server communication unit 33 communicates with the server 40 by applying a hole punching (Hole Punching) method to pass through the firewall 13 having a NAT. The hole punching method is a method for P2P communication between two peers blocked by NAT. That is, the server communication unit 33 transmits a UDP packet to the server 40 and forms a communication session of the server 40 using the IP address and port information of the transmitted packet.

In addition, the server communication unit 33 transmits the quality data to the server 40 or receives a call command from the server 40 using an SNMP trap.

SNMP communication is the communication between the server and the agent. In a quality measurement system, a server is a quality management system, and an agent is a probe (probe or measurement terminal), that is, an SNMP daemon installed in the probe. Therefore, the probe must have an SNMP daemon installed that can send and receive commands to and from the server.

In FIG. 8, the management station becomes a quality management system server and the management agent corresponds to a probe (probe or measurement terminal). Looking at the communication method between each other is as follows.

In general, when the server sends a get command to the probe, the SNMP agent of the probe sends the value of the probe corresponding to get to the server. For example, this command is used to know the status of probe A.

The Set command is used to set specific parameter values for the probe. A get or set is a synchronous feature that responds when requested between a server and a probe. For example, this command is used to set the measurement interval of probe A to 10 minutes. Alternatively, this function is used to command the server to measure quality from probe A to probe B.

Traps are used to send events generated by probes to the server asynchronously. For example, this function is used to send the quality measurement value that is automatically measured periodically between probe A and probe B.

In the absence of a firewall, there is a communication channel between the server and the probe, so you can start the above command from either the server or the probe. But you can't do that with a firewall.

NAT / FW function cannot start communication session from external network to internal network first, but if you start communication session from internal network to external network first, it depends on the keep alive time (NAT / FW). About 30 seconds) It is possible to communicate with the internal device (or probe) through the session opened from the corresponding port and the equipment (or quality control system) located in the external network.

Using this property, traps are sent from the probe inside NAT / FW to the external QC server at regular intervals. The server and probe can then make the desired SNMP communication via NAT / FW.

That is, the server may issue a command to the probe and receive a quality measurement value corresponding to the response or trap information from the probe.

The control unit 34 controls to receive a call command from the server 40 to form a telephone call or to transmit quality data to the server 40.

When the control unit 34 receives a call command from the server 40 through the server communication unit 33, the control unit 34 controls to connect the telephone call to the counterpart measurement terminal 30 through the call processing unit 31. At this time, the control unit 34 controls to measure the quality of the telephone call through the quality measuring unit 32 while the telephone call is connected.

In addition, the control unit 34 transmits the calculated quality data to the server 40 through the server communication unit 33, and collects and transmits the quality data periodically, or when one telephone call is terminated for the telephone call Send quality data.

In the example of FIG. 8, if the quality measurement between the originating probe A and the destination probe B is measured in one direction, the quality measurement result is the value measured at probe A, that is, the quality indicator value for the quality measurement for the media sent by probe B. Only will be collected. Only one measure of quality can be calculated for one way path.

If bi-directional measurement is possible under the same conditions, probe A may calculate a quality measurement for the media sent by probe B, and probe B may calculate a quality measurement with the media sent by probe A. In other words, the quality of the media can be measured in both directions with a single call, so the reliability of the result will be increased and efficient.

As mentioned above, although the invention made by this inventor was demonstrated concretely according to the said Example, this invention is not limited to the said Example and can be variously changed in the range which does not deviate from the summary.

11: IP Switch 12: Router
13: firewall 21a, 21b: internal network
22: Internet network 30: measuring terminal
40: server 50: database

Claims (7)

In a media quality measurement system in an encrypted VoIP network that measures the communication quality of Internet telephony between the internal network and the external network where the firewall is installed,
A plurality of measurement terminals installed in each of the plurality of internal networks, a plurality of measurement terminals that are given an Internet telephone number and have an Internet telephone call function, and calculate communication quality data when the telephone number is called. And,
A server for communicating with the measurement terminal through a Simple Network Management Protocol (SNMP) protocol, transmitting a call command to the measurement terminal or collecting quality data from the measurement terminal,
The media quality measurement system in an encrypted VoIP network, characterized in that the protocol of the Internet telephony call is encrypted.
The method of claim 1,
The server instructs a telephone call to the measurement terminal (hereinafter referred to as a transmission terminal) to a telephone number of a measurement terminal different from the measurement terminal (hereinafter referred to as a reception terminal) by scheduling. .
The method of claim 1, wherein the measurement terminal,
A call processor for forming a call path for an internet phone call and transmitting and receiving a test voice;
A quality measuring unit calculating quality data from the media including the test voice;
A server communication unit communicating with the server through an SNMP protocol; And,
And a control unit configured to receive a call command from the server to form a telephone call or to transmit the quality data to the server.
The method of claim 3,
And the server communication unit transmits the quality data to the server or receives a call command from the server using an SNMP trap.
The method of claim 3,
The quality measuring unit measures the quality of the media using a Real-time Transport Protocol (RTP).
The method of claim 5,
The quality measuring unit measures the quality of the media using a Real-time Transport Control Protocol (RTCP).
The method of claim 1,
The quality data is a media quality measurement system in an encrypted VoIP network comprising at least one of an R value, a MOS, a delay, jitter, a packet loss rate, and a call success rate. .

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