KR20100113853A - Adaptive codec switching method for vertical handover and sip client apparatus having adaptive codec switching function - Google Patents

Abstract

PURPOSE: An adaptive codec switching method for vertical handover and a sip client apparatus having an adaptive codec switching function are provided to supply high quality VoIP service for handover between heterogeneous networks by adaptively converting voice codec to a new network. CONSTITUTION: An apparatus measures a state value of a VoIP(Voice over Internet Protocol) session of a mobile terminal(S110), and calculates VoIP service quality based on the state value of the VoIP session(S120). The apparatus selects the codec of the most excellent VoIP service quality as the optimized codec in a codec group which calculates the VoIP service quality(S130). The apparatus converts the codec through a renegotiation process with the other terminals through the mobile terminal and a network(S140).

Description

SIP client device having adaptive codec conversion method and adaptive codec conversion function in heterogeneous network handover {ADAPTIVE CODEC SWITCHING METHOD FOR VERTICAL HANDOVER AND SIP CLIENT APPARATUS HAVING ADAPTIVE CODEC SWITCHING FUNCTION}

The present invention relates to a SIP client device having an adaptive codec conversion method and an adaptive codec conversion function during handover between heterogeneous networks, and more particularly, a network having a structure in which various heterogeneous networks are integrated and interworked based on IP (Internet Protocol). In case of handover between heterogeneous networks while a user is receiving Session Initiation Protocol (SIP) based Voice over Internet Protocol (VoIP) service, it provides high quality VoIP service by adaptively converting voice codec to network. It relates to a technique for doing.

Several methods have been proposed to support mobility of SIP-based VoIP services of UEs in heterogeneous networks with various characteristics.

First, there is a handover method using a multicast agent and a back-to-back SIP user agent (B2BUA). The B2BUA receives the SIP request from the mobile station, converts the SDP parameters, and sends it to the counterpart terminal. Each media stream is assigned a temporary multicast address, and the mobile station can provide a seamless service by sending a Re-INVITE message with a multicast address to the SIP B2BUA to transmit the stream over multiple networks.

Second, there is a make-before-break method. This is to reduce the handover delay time of a terminal by simultaneously connecting two or more networks. The method is to connect a new network first and then disconnect the existing network.

Finally, there is an adaptive codec change method, which not only reduces the handover delay time when the UE performs heterogeneous network handover, but also bit-rate and delay in conjunction with a real-time application in the application layer. And adaptive management and change of jitter.

However, the conventional adaptive codec changing method has a problem that no consideration is given to a method of selecting an appropriate VoIP codec according to network characteristics.

Therefore, the present invention is to solve the above problems of the prior art, when a handover between heterogeneous networks occurs while the user is receiving SIP-based VoIP service in a network of various heterogeneous networks integrated and interworking based on IP An adaptive codec conversion method for providing a high quality VoIP service by adaptively converting a voice codec to a network, and to provide a SIP client device having an adaptive codec conversion function.

An adaptive codec conversion method according to an aspect of the present invention for achieving the above object is a method for adaptively converting a codec to a new network during handover between mobile stations of a mobile terminal, the current VoIP session of the mobile terminal Measuring a state value of; Calculating a VoIP quality of service using the measured state value of the VoIP session; Selecting a codec having the highest VoIP quality of service as the optimal codec among the codec groups for which the VoIP quality of service is calculated; And converting a codec through a renegotiation process with the mobile station and a counterpart terminal connected through a network.

Measuring the state value of the current VoIP session of the mobile terminal is characterized by measuring the packet loss rate and delay time of the current VoIP session.

The state value of the current VoIP session is a packet loss rate and a delay time of the current VoIP session, and the packet loss rate and the delay time are measured using a Sender Report message of a Real-Time Transport Control Protocol (RTCP) protocol.

The delay time includes an uplink delay time, the uplink delay time,

Uplink delay time = when the other terminal sends Sender Report message to mobile terminal-When mobile terminal sends Sender Report message to other terminal-DLSR (Delay Since Last SR)

Is calculated according to.

In addition, the delay time includes a downlink delay time, the downlink delay time,

Downlink Latency = Round-trip time-Uplink Latency

Is calculated according to.

The calculating of the VoIP quality of service may be performed by calculating a quality of service using an ITU E-Model based R-Factor, and using an uplink and downlink R-Factor for a codec supported by the mobile terminal and the counterpart terminal. To calculate the quality of service.

The codec conversion may be performed through a renegotiation process with the counterpart terminal, after the mobile station transmits and receives two or more Sender Report messages with the counterpart terminal through a new network from the time when the handover of the mobile station is triggered. Is performed on.

On the other hand, the SIP client device having an adaptive codec conversion function according to another aspect of the present invention for achieving the above object, by detecting the handover of the mobile terminal equipped with the SIP client device by the handover A handover management module for triggering transmission of a SIP Re-INVITE message after acquiring the obtained new network address; A network delay prediction module for measuring a delay time of a current VoIP session; A packet loss rate measuring module for measuring a packet loss rate of a current VoIP session; And after calculating the VoIP service quality using the delay time and the packet loss rate measured by the network delay prediction module and the packet loss rate measurement module, the calculated VoIP service quality is the best among the codec groups for which the VoIP service quality is calculated. It includes an adaptive VoIP codec conversion module for converting the codec through the process of renegotiation between the mobile terminal and the other terminal connected via the network by selecting an excellent codec as an optimal codec.

The network delay prediction module and the packet loss rate measuring module measure delay time and packet loss rate using a Sender Report message of a Real-Time Transport Control Protocol (RTCP) protocol.

The delay time includes an uplink delay time, the uplink delay time,

Uplink delay time = when the other terminal sends Sender Report message to mobile terminal-When mobile terminal sends Sender Report message to other terminal-DLSR (Delay Since Last SR)

Is calculated according to.

In addition, the delay time includes a downlink delay time, the downlink delay time,

Downlink Latency = Round-trip time-Uplink Latency

Is calculated according to.

The adaptive VoIP codec conversion module calculates the VoIP quality of service using an ITU E-Model based R-Factor and performs uplink and downlink R-Factor for the codecs supported by the mobile terminal and the counterpart terminal. To calculate the quality of service.

In addition, the adaptive VoIP codec conversion module, after the mobile terminal transmits two or more Sender Report messages with the opposite terminal through a new network from the time when the handover of the mobile terminal is triggered, The renegotiation process converts the codec.

According to the present invention, when a handover between heterogeneous networks occurs, the voice codec is adaptively converted to a new network, thereby providing a high-quality VoIP service even during handover between heterogeneous networks.

Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings so that those skilled in the art may easily implement the present invention. However, in describing the preferred embodiment of the present invention in detail, if it is determined that the detailed description of the related known function or configuration may unnecessarily obscure the subject matter of the present invention, the detailed description thereof will be omitted. In addition, the same reference numerals are used throughout the drawings for parts having similar functions and functions.

In addition, in the entire specification, when a part is referred to as being 'connected' to another part, it may be referred to as 'indirectly connected' not only with 'directly connected' . Also, to "include" an element means that it may include other elements, rather than excluding other elements, unless specifically stated otherwise.

In addition, the term 'module' refers to a unit for performing a specific function or operation, which may be implemented in hardware or software or a combination of hardware and software.

The present invention proposes a method in which a terminal receiving SIP-based VoIP service adaptively selects and converts a voice codec to a network during heterogeneous network handover in a situation where various networks having different characteristics coexist, such as next-generation networks. .

In the case of handover between heterogeneous networks, since the characteristics of the network before and after the handover are different, the quality of the VoIP service may appear when the voice data is transmitted using the same codec as before even in the new network. Therefore, in case of inter-network handover, a voice codec suitable for the characteristics of the new network should be selected, which is called adaptive codec conversion.

1 is a diagram illustrating a network to which an adaptive codec conversion method according to an embodiment of the present invention is applied.

For example, as shown in FIG. 1, the present invention may be applied to a network in which various networks coexist, such as a UMTS network, a WiBro network, and a WLAN network. The mobile terminal of the user has a connection with several networks having different characteristics, and among them, a VoIP service can be received through a specific network. At this time, the SIP protocol is used for the VoIP session.

2 is a diagram illustrating a renegotiation process when handing over from a UMTS network to a WLAN network according to an embodiment of the present invention.

When a user is receiving VoIP service through a UMTS network (S10) and a handover occurs to the WLAN network due to a handover policy, the mobile station transmits a SIP Re-INVITE message to itself and a counterpart terminal connected through the network. A handover occurs indicating that the session needs to be updated. After receiving the Re-INVITE message, the counterpart terminal updates the session information, sends a SIP OK message to the mobile station, and starts transmitting RTP (Real-Time Transport Protocol) data to the new network address of the mobile station. Meanwhile, after receiving the SIP OK message, the mobile station sends a SIP ACK message to the counterpart terminal and starts transmitting the RTP data through the new network (S20 and S30).

Thereafter, the RTP protocol is basically performed together with the Real-Time Transport Control Protocol (RTCP), and the RTCP periodically checks the state of the RTP session of the terminal using a Sender Report message and informs the opposite terminal (S40).

Thereafter, the mobile station performs codec conversion by transmitting a SIP Re-INVITE message to the counterpart terminal after a predetermined time T has elapsed from the time when the handover is triggered (S50). Here, the predetermined time T means a time at which the mobile terminal can exchange two or more Sender Report messages with the counterpart terminal through a new network.

4 is a flowchart of an adaptive codec conversion process according to an aspect of the present invention.

Adaptive codec conversion method according to an aspect of the present invention, the current VoIP session state measurement step (S110), VoIP service quality calculation step (S120), the optimal codec selection step (S130) and through the renegotiation process with the other terminal A codec conversion step S140 is included.

In the current VoIP session state measurement step (S110), the state value of the current VoIP session of the mobile station, that is, packet loss rate and delay time, is measured. This can be obtained by using the Sender Report message of the RTCP protocol.

The NTP Time Stamp field included in the Sender Report message indicates when the corresponding Sender Report message is transmitted, and the Delay Since Last SR (DLSR) field indicates when the terminal that sent the Sender Report message last received the Sender Report message. This is the time between sending the corresponding Sender Report message from. In addition, the Fraction Lost field indicates the degree of loss of a packet generated between the time of transmitting the previous Sender Report message and the time of transmitting the current Sender Report message. Since every RTP data packet is assigned a sequence number, it is possible to count the number of packets that have not arrived. In the Fraction Lost field, information on the ratio of such lost packets is carried. In the present invention, this value can be used as a packet loss rate.

On the other hand, each terminal stores the time when it last sent the Sender Report message. Accordingly, when the terminal receives the Sender Report message from the counterpart terminal, a round trip time (RTT) may be calculated according to the following equation.

RTT = When the mobile station receives the sender report message from the other terminal-When the mobile station last sends the sender report message-DLSR

In addition, using the RTT calculated as described above, the delay time of the uplink and the downlink can be calculated according to the following equation.

Uplink delay time = when the other terminal sends a Sender Report message to the mobile station-when the mobile terminal sends a Sender Report message to the other terminal-DLSR

Downlink Latency = RTT-Uplink Latency

Then, in the VoIP quality of service calculation step (S120), VoIP quality of service is calculated using an RTU-Factor based on ITU E-Model. The quality of service is calculated by using uplink and downlink R-Factor for the codec supported by both mobile station and counterpart, and R-Factor is calculated according to the following equation.

Here, I _d means the degree of damage due to the delay time, I _e means the degree of damage due to packet loss. These are respectively calculated according to the following formula.

이고, d는 msec 단위의 지연시간이다.here,

D is a delay time in msec.

Here, λ ₁ is a codec-dependent impairment factor and λ ₂ is a codec dependent factor robust to the packet loss. In addition, PLR (Packet loss rate) is a packet loss rate.

Then, in the optimal codec selection step (S130), the optimal codec is selected in the new network by using the R-Factor calculated for each codec, that is, the VoIP service quality. In other words, based on the VoIP service quality calculated for each codec supported by both the mobile terminal and the counterpart terminal, the codec having the best VoIP service quality is selected as the optimal codec in the new network.

Subsequently, in the codec conversion step S140, the mobile station transmits a Re-INVITE message containing a new codec list to the counterpart terminal, thereby converting the codec through a renegotiation process with the counterpart terminal.

5 is a block diagram of a SIP client device having an adaptive codec conversion function according to another aspect of the present invention.

The SIP client device according to another aspect of the present invention is mounted on a terminal for adaptive codec conversion, and an additional configuration for performing adaptive codec conversion is required in addition to the basic SIP client for SIP-based services. Hereinafter, description will be given focusing on an additional configuration for adaptive codec conversion.

The SIP client device 200 should further include a handover management module 210, a network delay prediction module 220, a packet loss rate measuring module 230, and an adaptive VoIP codec conversion module 240.

The handover management module 210 detects the handover of the terminal on which the SIP client device 200 is mounted, acquires a new network address obtained by the handover by the terminal, and uses the same to provide the adaptive VoIP codec conversion module 240. Triggers the transmission of a SIP Re-INVITE message.

The network delay prediction module 220 measures the delay time of the current VoIP session. The network delay prediction module 220 calculates uplink and downlink delay times by using a Sender Report message transmitted and received by the terminal.

The packet loss rate measuring module 230 measures the packet loss rate of the current VoIP session.

The adaptive VoIP codec conversion module 240 calculates the VoIP quality of service using the delay time and the packet loss rate measured by the network delay prediction module 220 and the packet loss rate measurement module 230, and according to the result, After the codec is selected, the codec is converted through renegotiation with the other terminal.

Delay time measurement by the network delay prediction module 220, packet loss rate measurement by the packet loss rate measurement module 230, and service quality calculation by the adaptive VoIP codec conversion module 240, optimal codec selection and codec conversion As described above with reference to FIG. 4, a detailed description thereof will be omitted.

The present invention is not limited by the above-described embodiment and the accompanying drawings. It will be apparent to those skilled in the art that the present invention may be substituted, modified, and changed in accordance with the present invention without departing from the spirit of the present invention.

1 is a diagram illustrating a network to which an adaptive codec conversion method according to an embodiment of the present invention is applied;

2 is a diagram illustrating a renegotiation process when handing over from a UMTS network to a WLAN network according to an embodiment of the present invention;

3 is a diagram for explaining a time point at which adaptive codec conversion is performed according to the present invention;

4 is a flowchart of an adaptive codec conversion process according to an aspect of the present invention;

5 is a block diagram of a SIP client device having an adaptive codec conversion function according to another aspect of the present invention.

Claims (15)

In the method of adaptively converting a codec to a new network during the handover between mobile devices of the heterogeneous network, Measuring a state value of a current VoIP session of the mobile terminal; Calculating a VoIP quality of service using the measured state value of the VoIP session; Selecting a codec having the highest VoIP quality of service as the optimal codec among the codec groups for which the VoIP quality of service is calculated; And And converting a codec through a renegotiation process between the mobile terminal and a counterpart terminal connected through a network. The method of claim 1, Measuring the status value of the current VoIP session of the mobile terminal is adaptive codec conversion method for heterogeneous network handover, characterized in that for measuring the packet loss rate and delay time of the current VoIP session. The method of claim 1, The state value of the current VoIP session is a packet loss rate and a delay time of the current VoIP session, and the measurement of the packet loss rate and the delay time is performed using a Sender Report message of a Real-Time Transport Control Protocol (RTCP) protocol. Adaptive codec conversion method for handover between heterogeneous networks. The method according to claim 2 or 3, The delay time includes an uplink delay time, the uplink delay time, Uplink delay time = when the other terminal sends Sender Report message to mobile terminal-When mobile terminal sends Sender Report message to other terminal-DLSR (Delay Since Last SR) Adaptive codec conversion method for handover between heterogeneous networks, characterized in that calculated according to. The method according to claim 2 or 3, The delay time includes a downlink delay time, the downlink delay time, Downlink Latency = Round-trip time-Uplink Latency Adaptive codec conversion method for heterogeneous handover, characterized in that calculated according to. The method of claim 1, The step of calculating the VoIP quality of service is an adaptive codec conversion method for handover between heterogeneous networks, characterized in that for calculating the quality of service using the R-Factor based on ITU E-Model. The method of claim 1, The calculating of the VoIP quality of service is adaptive for handover between heterogeneous networks, characterized by calculating quality of service using uplink and downlink R-Factor for the codec supported by the mobile terminal and the counterpart terminal. Codec conversion method. The method of claim 1, The codec conversion may be performed through a renegotiation process with the counterpart terminal, after the mobile station transmits and receives two or more Sender Report messages with the counterpart terminal through a new network from the time when the handover of the mobile station is triggered. Adaptive codec conversion method for hand-over between heterogeneous networks, characterized in that performed in. A handover management module for detecting a handover of a mobile terminal equipped with a SIP client device to trigger transmission of a SIP Re-INVITE message after the mobile terminal obtains a new network address obtained by handover; A network delay prediction module for measuring a delay time of a current VoIP session; A packet loss rate measuring module for measuring a packet loss rate of a current VoIP session; And After calculating the VoIP service quality by using the delay time and the packet loss rate measured by the network delay prediction module and the packet loss rate measurement module, the calculated VoIP service quality is the highest among the codec groups for which the VoIP service quality is calculated. And an adaptive VoIP codec conversion module for selecting a codec as an optimal codec and converting the codec through a renegotiation process between the mobile terminal and a counterpart terminal connected through a network. The method of claim 9, The network delay prediction module and the packet loss rate measuring module measure delay time and packet loss rate using a Sender Report message of a Real-Time Transport Control Protocol (RTCP) protocol. The method of claim 10, The delay time includes an uplink delay time, the uplink delay time, Uplink delay time = when the other terminal sends Sender Report message to mobile terminal-When mobile terminal sends Sender Report message to other terminal-DLSR (Delay Since Last SR) SIP client device, characterized in that calculated according to. The method of claim 10, The delay time includes a downlink delay time, the downlink delay time, Downlink Latency = Round-trip time-Uplink Latency SIP client device, characterized in that calculated according to. The method of claim 9, The adaptive VoIP codec conversion module is a SIP client device, characterized in that for calculating the VoIP quality of service using an ITU E-Model based R-Factor. The method of claim 9, The adaptive VoIP codec conversion module is configured to calculate the quality of service using uplink and downlink R-Factor for the codec supported by the mobile terminal and the counterpart terminal. The method of claim 9, The adaptive VoIP codec conversion module performs a renegotiation process with the counterpart after the mobile station transmits two or more Sender Report messages with the counterpart through a new network from the time when the handover of the mobile station is triggered. SIP client device characterized in that for converting the codec through.

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