KR100378785B1 - Method for receiving voice using streaming type - Google Patents

Abstract

A voice receiving method using a streaming method is disclosed. The present invention provides a system for processing voice as data in a mobile communication network, comprising: a) storing a packet of a voice signal generated in an active section of a voice signal for a predetermined time (Tb) continuously in a receiving buffer of a receiving terminal; step; b) outputting packets one by one when the predetermined time Tb of step a) has elapsed, and inputting one input packet per output into the reception buffer if there is an input packet; c) determining whether there are residual packets in the reception buffer; d) determining that the active period ends if there is no remaining packet as a result of the determination of step c); And e) stopping the use of the reception buffer if the active period is terminated as a result of the determination of step d). Therefore, according to the present invention, it is possible to reduce the probability that the voice is interrupted during the active period.

Description

Method for receiving voice using streaming type {Method for receiving voice using streaming type}

An object of the present invention is to provide an efficient streaming method of receiving a voice centering on an active section of a voice at a receiving end of a VoIP service applied to a mobile communication system.

Voice over Internet Protocol (VoIP) is a term for Internet Protocol (IP) telephony technology for a series of facilities that deliver voice information using the Internet Protocol (IP). In general, this means that voice information is sent in digital form within discrete packets, rather than a traditional protocol based on the circuit, such as PSTN, a public switched telephone network. The main advantage of VoIP and Internet telephony technology is that telephony services can be integrated using existing IP networks, allowing telephony users to receive long-distance and international telephony services over the Internet and intranet environments with only local telephone charges.

Voice over IP (VoIP) forums are designed to encourage the use of the International Telecommunication Union (TTU) -T H.323, a standard for transmitting voice (sound) and video signals over IP on the public Internet or intranets within an enterprise. It was made by major equipment manufacturers such as Vocaltech, 3 Com, and NetSpeak. The VoIP Forum also encouraged service standards, allowing users to locate other users and enabling the use of touch phone signals for automatic telephone distribution and voice mail.

However, when applying a Voice over IP (VoIP) service to a mobile communication system, a data link layer protocol used in a wireless section with jitter, that is, a change in a time delay value of packets according to the use of an IP network, In the CDMA cellular system, retransmission occurs using Radio Link Protocol, that is, RLP protocol. When a retransmission occurs, the packet and the next packet are received later, and the number of stored packets in the buffer is reduced by that amount. Therefore, when retransmission occurs more than a certain number, the buffer becomes empty. In this case, the decoding and reproducing of the received voice are stopped until the buffer is refilled, thereby degrading the voice quality.

The present invention has been invented to solve the above problems, and an object of the present invention is to refill the buffer with a predetermined number of packets at the start of the active section of the received voice signal, thereby stopping the voice playback during the active section. It is to reduce the probability of becoming.

In order to achieve the above object, the present invention provides a system for processing voice as data in a mobile communication network, comprising: a) voice generated in an active section of a voice signal for a predetermined time (Tb) continuously in a reception buffer of a receiving terminal; Storing a packet of signals; b) outputting packets one by one when the predetermined time Tb of step a) has elapsed, and inputting one input packet per output into the reception buffer if there is an input packet; c) determining whether there are residual packets in the reception buffer; d) determining that the active period ends if there is no remaining packet as a result of the determination of step c); And e) stopping the use of the reception buffer if the active period is terminated as a result of the determination of step d).

Therefore, according to the present invention, it is possible to reduce the probability that the voice is interrupted during the active period.

1 is a flowchart illustrating a voice receiving method according to the present invention.

2 illustrates an operation process of the present invention.

<Description of the symbols for the main parts of the drawings>

1: voice active section start point 2: voice active section end point

3: Voice active section 4: Voice inactive section

Hereinafter, with reference to the accompanying drawings will be described a preferred embodiment of the present invention.

1 is a flowchart illustrating receiving a voice packet in a streaming manner in a receiving terminal according to the present invention. The present invention operates the reception buffer only in the active section of the voice. When the active section of the voice starts (S10), the packet of the voice signal is continuously stored in the reception buffer of the receiver side (S20).

The number of packets of the voice signal that can be stored in the buffer can be arbitrarily set, and in general, if the time Tb has elapsed enough to fill the capacity of the voice signal that can be stored in the buffer (S30), The voice is decoded by outputting the packets of the voice signal one by one from the buffer in accordance with the transmission period of the voice packet (20 msec in the case of CDMA cellular) (S40).

However, if time Tb has not elapsed enough to fill the buffer (S30), step S20 is performed.

In performing the step S40, if there is an input packet to be input to the buffer (S42), one input packet for one output is input to the receiving buffer (S44). That is, when one packet is decoded and output in the buffer, a new input packet is stored in the empty position. If the input packet does not exist, it is determined whether there are remaining packets remaining in the buffer (S50).

If there are no packets of the remaining voice signal remaining in the buffer (S50), it is determined whether the active section of the voice is terminated (S60). The step S40 must be performed until there are no packets of the remaining voice signals remaining in the buffer.

As a result of the determination in step S60, when the active section of the voice is terminated and the inactive section starts, the use of the reception buffer is stopped (S70).

However, at the beginning of the next active period, the buffer is reset, performing the steps described above. This refills a certain number of packets in the buffer, during which no packets are output from the buffer.

2 illustrates an operation process of the present invention.

In the actual voice communication, the voice active section 3 and the voice inactive section 4 appear alternately, and in general, the voice active section 3 of the first caller becomes the voice inactive section 40 of the second caller. The voice inactive section 4 of one caller becomes the voice active section 30 of the second caller.

In the voice inactive period (4, 40), even if the voice packet is not received, there is no significant difference in sound quality. Voice decoding is not interrupted when retransmission occurs less than the buffer acceptance limit for the receive buffer.

When the voice signal active section 3 starts, N consecutive packets are stored in the receiving buffer of the receiving terminal (in the normal state, N packets can be maintained in the buffer afterwards).

The buffer is not operated in the voice inactive section 4 generated after the voice active section ending point 2. That is, if the signal is terminated by checking whether the active section of the signal has ended (inactive section starts), the packet is decoded after receiving all packets remaining in the reception buffer without receiving the packet during the next inactive section. Stop reception.

However, if the voice active section start point 1 is detected, the buffer is operated again.

Embodiment of the present invention is only one embodiment, many modifications and variations of the components of the present invention can be made within the scope not departing from the gist of the invention, of course, the present invention is not limited to the embodiment. .

As described in detail above, in the voice reception method using the streaming method of the present invention, the reception buffer is operated only in the active section, and the buffer is empty during the active section by resetting the buffer at the beginning of the next active section. By reducing the probability of this temporary interruption, the received voice quality can be improved. In addition, when the reception buffer is operated only in the active period, the probability that the buffer is vacated in the active period is lower compared to the case where the buffer is operated in the inactive period, and thus the probability of the voice decoding and reproduction being stopped is also lower.

Claims (2)

In a system for processing voice as data in a mobile communication network, a) storing a packet of a voice signal generated in an active section of the voice signal for a predetermined time Tb continuously in a reception buffer of the receiving terminal; b) outputting packets one by one when the predetermined time Tb of step a) has elapsed, and inputting one input packet per output into the reception buffer if there is an input packet; c) determining whether there are residual packets in the reception buffer; d) determining that the active period ends if there is no remaining packet as a result of the determination of step c); And e) stopping the use of the reception buffer if the active period is terminated as a result of the determination of step d). The voice receiving method of claim 1, wherein when the active section is not terminated as a result of the determination of step d), steps a) to e) are performed.