1. Field of the Invention
The present invention relates to an apparatus and method for processing a packet in a voice and data integration system.
2. Description of the Related Art
Owing to today's rapid popularization of Internet and its accompanying request for various services, an Internet protocol (IP) network is being remarkably advanced in its performance and service. Accordingly, a request for more various services is continuously increased.
IP network-based voice transmission, one of such services, plays a great role in the IP network together with data transmission. Hence, a request for various voice transmission functions is also made together. Accordingly, terminals such as a digital telephone and a single telephone have been required to integrate voice IPs (VoIP: Voice over Internet Protocol).
Thus, a voice and data integration apparatus begins to be developed so that conventional terminals and Internet terminals can exchange data or voice with each other through the IP network.
Such a voice and data integration apparatus performs a gateway function of matching the IP network and each terminal, and performs a VoIP gateway function of decoding a voice packet received from the IP network into pulse code modulation (PCM) data and transmitting the decoded data to the terminal, or encoding the PCM data received from the terminal into the voice packet and transmitting the encoded data to the IP network.
Further, main factors deteriorating a quality of voice communication in a network such as an IP network include a loss of packet, a delay, and a jitter.
1) Loss of Packet: phenomenon in which a packet is lost during transmission on the network and thus not decoded in the voice and data integration apparatus, and thereby a quality of voice is remarkably deteriorated.
2) Delay: phenomenon in which a packet is temporally delayed when output, compared to when input, and each packet is differently delayed depending on a route along which the packet is transmitted on the network.
3) Jitter: phenomenon in which queuing and congestion occurring in network equipment due to a different rate of delay of each packet. Each piece of network equipment should include a jitter buffer in order to prevent this phenomenon, but the jitter buffer causes transmission delay.
Accordingly, in order to guarantee the quality of voice communication over the IP network, techniques such as dynamic jitter buffering for dynamically adjusting a size of the jitter buffer, Real-time Transport Protocol (RTP) compressing for compressing a Real-time Transport Protocol (RTP) packet as a voice packet, and Real-time Transport Protocol (RTP) multi-framing for splitting a voice packet have been utilized. See Network Working Group Request for Comments (RFC) 1889-RTP: A Transport Protocol for Real-Time Applications; Audio-Video Transport Working Group, H. Schulzrinne et al. January 1996 or Network Working Group Request for Comments (RFC) 3550-RTP: A Transport Protocol for Real-Time Applications; H. Schulzrinne et al. July 2003. However, a perfect quality of voice is not yet guaranteed.
Further, in the conventional voice and data integration apparatus, an algorithm for guaranteeing the quality of voice is processed only in a digital signal processing (DSP) manufactured in a manufacture company. Hence, the algorithm is not so good in extensibility and reliability.