Telephonic communications – Audio message storage – retrieval – or synthesis – Interaction with an external nontelephone network
Reexamination Certificate
2001-09-04
2004-09-21
Tsang, Fan (Department: 2645)
Telephonic communications
Audio message storage, retrieval, or synthesis
Interaction with an external nontelephone network
C370S252000
Reexamination Certificate
active
06795534
ABSTRACT:
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to a data recording technique for the so-called IP or Internet telephony. More particularly, the invention relates to a data recording system for IP telephony that realizes approximately the same voice or speech communication as the conventional, ordinary telephony by the use of the Internet Protocol (IP) and the IP-based computer network.
With IP telephony, voice or audio data (i.e., audio data) to be transmitted is divided into IP packets (i.e., audio IP packets) and then, these packets are successively sent from a telephone terminal to another distant from it by way of an IP-based computer network. When audio data is transmitted using IP telephony, audio IP packets corresponding to the audio data need to be processed in real time. On the other hand, if audio data is transmitted as it is, it is impossible to ensure a required frequency bandwidth for data transmission. This is because the IP based network is of the so-called “best effort” type. As a result, conventionally, audio data is usually transmitted using the User Datagram Protocol (UDP) and the Realtime Transport Protocol (RTP) for the transport layer of the well-known OSI (Open Systems Interconnection) reference model.
With IP telephony, voice or audio data (i.e., audio data) to be transmitted is divided into IP packets (i.e., audio IP packets) and then, these packets are successively sent from a telephone terminal to another distant from it by way of an IP-based computer network. When audio data is transmitted using IP telephony, audio IP packets corresponding to the audio data need to be processed in real time. On the other hand, if audio data is transmitted as it is, it is impossible to ensure a required frequency bandwidth for data transmission. This is because the Ip-based network is of the so-called “best effort” type. As a result, conventionally, audio data is usually transmitted using the User Datagram Protocol (UDP) and the Realtime Transport Protocol (RTP) for the transport layer of the well-known OSI (Open Systems Interconnection) reference model.
Since IP telephony uses the IP protocol for data transmission, part of the IP packets tend to be lost during transmission and the packets thus lost are automatically resent from the telephone terminal from which the lost IP packets are originated. Generally, the loss rate of the IP packets during transmission varies dependent on the current amount of the traffic on an IP-based network. Thus, there is a problem the voice or speech communication quality is likely to deteriorate.
Moreover, the amount of the traffic on an IP network fluctuates at all times and abrupt increase of the traffic amount is unable to be anticipated. If the fluctuation of the traffic amount can be controlled to an extent by a process such as giving the order of priority to the audio IP packets, the above-described problem about the communication quality deterioration may be suppressed. In this case, however, not only the functions of routers connected to the network but also the entire operation of the network itself need to be additionally controlled. Thus, it is not realistic.
Additionally, even if only the operation of the IP telephone terminal is controlled to give the order of priority to the audio IP packets, it is difficult to make sure that the audio data packets are transmitted through the network as intended.
Furthermore, if the loss rate of the audio packets increases when the amount of traffic is large, the packet retransmission process and/or the congestion control process is/are not performed, where only the real-time process using the RTP protocol is carried out. Therefore, in this case, the speech quality deterioration becomes more conspicuous.
Considering the above-described characteristics of IP telephony, conventionally, a function to complement the lost IP packets during transmission is incorporated into the IP telephone terminal and/or the IP telephone subscriber circuit of an exchange. This lost-packet complementing function is implemented by anticipating the audio data contained in a lost IP packet based on its precedent and subsequent audio data. Therefore, the audio data complemented by this function does not accord perfectly with that contained in the lost packet. Although various researches on the lost-packet complementing function have been conducted, it is unable to realize complete reproduction of the original voice or speech as long as this function is used.
The complement of the lost packet is more difficult if the loss rate of the packets increases furthermore as the traffic increases. This means that in this case, the complemented data is noticeably different from the original one. Thus, the reproduced speech tends to include some sensible distortion.
As explained above, when audio data is transmitted over the IP-based network, the quality of the reproduced speech is affected by the fluctuation of uncontrollable traffic. Thus, to keep the quality degradation and the transmission delay of audio data over the IP network at the same level as the conventional, ordinary telephony by way of the telephone lines, a control method using the TOS (Type of Service) field included in the header of an IP packet may be adopted, for example. However, in this method, high performance routers capable of interpreting the content of the TOS field are required over the whole IP-based network. As a result, this method is difficult to be adopted practically.
Moreover, a voice or sound recording method on a magnetic tape or the like (which has been incorporated in the conventional telephone-answering machines) may be adopted to realize approximately the same quality degradation and approximately the same transmission delay over the IP-based network as the conventional, ordinary telephony. Ordinary telephone-answering machines record directly the voice or speech generated by the handset of a telephone on a magnetic tape. If this direct recording method in the ordinary answering machine is applied to the IP telephone terminal, there is a possibility that the quality of reproduced voice or speech tends to degrade due to the loss of IP packets described previously. As a result, even if the voice or speech generated by the IP telephone terminal is directly recorded on a magnetic tape, the above-identified disadvantage of IP telephony (i.e., the complete reproduction of original voice or speech is impossible) is unable to be solved.
SUMMARY OF THE INVENTION
Accordingly, an object of the present invention is to provide a data recording system for IP telephony that complements perfectly the lost audio data due to loss of the IP packets during transmission.
Another object of the present invention is to provide a data recording system for IP telephony that eliminates the necessity of the lost-packet complementing function for the lost audio IP packets.
The above objects together with others not specifically mentioned will become clear to those skilled in the art from the following description.
A data recording system for IP (Internet Protocol) telephony according to the invention comprises;
(a) an IP-based network;
(b) a first telephone terminal connected to the network;
the first telephone terminal being capable of transmission and reception of audio data in the form of IP packets, making communication using an IP;
(c) a second telephone terminal connectable directly to the network or indirectly thereto by way of an exchange;
the second telephone terminal being capable of speech communication; and
(d) a recording device connected to the network;
the recording device being capable of recording audio data transmitted between the first telephone terminal to the second telephone terminal;
wherein when communication is performed between the first telephone terminal and the second telephone terminal, speech IP packets corresponding to audio data are formed and then, the speech IP packets thus formed are transmitted between the first telephone terminal and the second telephone terminal by way of the network in app
Gauthier Gerald
Tsang Fan
Young & Thompson
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