KR20000034514A - Method for transmitting audio data of a real time transport protocol - Google Patents

Abstract

PURPOSE: A method for transmitting audio data of a real time transport protocol is provided so that the deterioration of the voice quality causing by the lack of the network source can be prevented by transmitting the RTP packet added UDP and IP headers, thereby being removed the repetitive transmission of the UDP and IP headers. CONSTITUTION: A method for transmitting audio data of a real time transport protocol includes several steps. A step is to construct RTP packet by adding and multiplexing an RTP header to each audio data generated from the large number of audio sources. A step is to construct UDP packet by adding UDP header to the RTP packet. A step is to add an IP header to the UDP packet and transmit it.

Description

A method for transmitting a real-time transport protocol audio data

The present invention relates to a method for transmitting real-time transport protocol (RTP) audio data, and in particular, an IP (Internet Protocol) header and a User Datagram Protocol (UDP) that overlap each packet when transmitting audio data through RTP. The present invention relates to a method of transmitting RTP audio data that can overcome the inefficiency of network resources by a header.

The Internet is an interworking network that enables the transmission of information between various networks of various technologies and devices using a protocol aggregation called TCP / IP. With the openness and compatibility of the TCP / IP protocol, the Internet has grown into a global network, with subscribers growing very rapidly, especially in academic and research institutions. In addition, as the development of the World Wide Web (WWW) technology and the public's interest have increased, the Internet has attracted the attention of the mass media and the industry as the leaders of the information age.

However, current Internet-based technologies lack so much in terms of functionality, efficiency, security, scalability, and flexibility, in order to handle new and diverse application services as reliably and efficiently as users want in a huge and complex network environment at a very high speed. Therefore, the development of new Internet technology to overcome this is actively progressing.

One of the next generation technologies of the Internet, "RTP", was developed in response to the above demands. RTP is an end-to-end transmission service function required for an application that transmits real-time information such as interactive audio, video, and simulation data. Protocol to provide.

Services provided by the RTP include data type identification, checking of sequence numbers, internal timestamp delivery, and monitoring of data transmission. However, RTP itself does not guarantee real-time transmission, guarantee of transmission order, or other forms of quality of service. It simply passes information through the RTP packet so that the receiver can reconstruct the data. That is, RTP has a structure that modifies and complements the functions of the underlying network in terms of application programs rather than performing all the functions of the transport layer. In addition, RTP does not exist as a separate layer, but rather is comprehensively included in the processing portion of an application.

In general, an application program executes RTP over UDP in order to take advantage of the multi-transmission and checksum functions provided by UDP. In other words, UDP and RTP protocols are combined to support transport layer services required by applications.

In addition, RTCP (Real-time Transport Control Protocol), which is a control protocol in addition to RTP, is involved in data transmission to monitor data transmission and provide minimal control and authentication. The RTCP is a control packet periodically sent by every participant in the session, the function of which provides feedback on the quality of service (QoS) of the transmitted data, and carries a DNAME, which is the transport layer ID for the RTP source. Since the number of participants can be known from these RTCP packets, the transmission speed can be controlled.

The above-described RTP can be applied to a videoconferencing system in which end-to-end audio data and video data are separated and transmitted using different UDP ports, and only two DNAMEs in RTCP are related to each other to associate the two streams. Synchronization can be matched with the timestamp of the RTP basic header.

On the other hand, conventionally, the RTP packet shown in FIG. 1 was used to transmit audio data through RTP. Each RTP packet consists of audio data captured from one sound source and a corresponding RTP header. When transmitting an RTP packet to a counterpart via the Internet, a UDP header and an IP header are added to each RTP packet in order.

However, as in the prior art, if the same UDP header and IP header are added and transmitted for each RTP packet composed of audio data and RTP header captured from one sound source, the amount of unnecessary traffic increases, resulting in efficient use of network resources. There was no problem.

That is, the size of the UDP header and the IP header is 28 bytes, which is unlikely to have a significant effect on the traffic. However, since the RTP packets are transmitted about once every 20 ms, the cumulative amount is very large.

SUMMARY OF THE INVENTION The present invention has been made to solve the above problems, and adds an RTP header to each of audio data simultaneously captured from a plurality of sound sources, and then multiplexes to form one RTP packet, and finally UDP to the configured RTP packet. It is an object of the present invention to provide a method of transmitting RTP audio data that can overcome the inefficiency of network resources due to repeated transmission of a UDP header and an IP header by adding a header and an IP header.

In order to achieve the above object, a method of transmitting RTP audio data according to the present invention comprises the steps of adding an RTP header to each of audio data captured from a plurality of sound sources, and then multiplexing them to form an RTP packet; Adding a UDP header to configure a UDP packet, and adding and transmitting an IP header to the UDP packet.

1 is a block diagram of a transport packet according to the prior art,

2 is a block diagram of a transport packet according to an embodiment of the present invention;

3 is a format of a Real-time Transport Protocol (RTP) header applied to the present invention.

Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings.

2 is a configuration diagram of a transport packet according to an embodiment of the present invention, wherein the transport packet includes an IP header, a UDP header, an RTP header of n sound sources, and audio data. Here, the format of each RTP header is shown in FIG.

In FIG. 3, the V (Version) field is a 2-bit field defining a version of the RTP, the P (Padding) field is a 1-bit field defining whether a padding octet is included, and the X (eXtension) field has a header extension. CC (CSRC count) field is a 4-bit field containing the number of CSRC (Contributing SouRCe) identifiers that are IDs of each sound source, and M (Marker) field is a 1-bit field defined by the profile. to be.

In addition, the PT (Payload Type) field is a 7-bit field that defines the format of the RTP payload, and the Seq # (Sequence number) field is a 16-bit field that is increased by 1 for each RTP data packet transmission. Used to detect loss and recover packet sequences, the timestamp field is a 32-bit field that defines timestamp information that periodically samples the reference timer, and the SSRC (Synchronization SouRCe) identifier field is a 32-bit field that defines the synchronization source. The CSRC identifier field is a 32-bit field that defines the sound source for the payload included in the packet, and can define 0 to 15 items, and 32 bits are allocated to each item.

Details of the RTP header described above are described in the RFC 1889, 1890 specification book (Internet homepage http://sunsite.auc.dk/RFC/rfc/rfc1889.html, http://sunsite.auc.dk /RFC/rfc/rfc1899.html).

A method of transmitting RTP audio data according to an embodiment of the present invention using a transport packet configured as described above is as follows.

First, audio data 1, audio data 2,... Captured simultaneously from n sound sources. , RTP header 1, RTP header 2,... RTP header n is added and then multiplexed to form an RTP packet. Here, RTP header 1, RTP header 2,... Added to each audio data. , RTP header n is configured in the format shown in FIG. 3 according to the audio data.

Subsequently, a UDP packet is configured by adding a UDP header to the RTP packet configured as described above, and an IP header is added to the UDP packet and transmitted to the other party through the Internet. Here, one UDP packet includes a plurality of RTP packets.

That is, conventionally, a UDP header and an IP header are added and transmitted for each RTP packet including one audio data and one RTP header, whereas an embodiment of the present invention provides an RTP packet including n audio data and n RTP headers. The UDP header and the IP header are added to the IP header, and when considered from a different angle, conventionally, n UDP headers and IP headers must be transmitted in order to transmit audio data captured from n sound sources. The example only needs to send one UDP header and one IP header, resulting in a significant traffic reduction.

As described above, since the present invention transmits by adding a UDP header and an IP header to an RTP packet including audio data and corresponding RTP headers that are simultaneously generated from a plurality of sound sources, repeated transmission of the UDP header and the IP header is eliminated, thereby reducing network resources. This saves, and furthermore, there is an effect that can prevent the sound quality degradation caused when the network resources are insufficient.

Claims (1)

Adding RTP headers to each of audio data captured from a plurality of sound sources, and then multiplexing them to construct an RTP packet; Configuring a UDP packet by adding a UDP header to the RTP packet; RTP audio data transmission method comprising the step of transmitting by adding an IP header to the UDP packet.