Data gateway and method for relaying data

Multiplex communications – Pathfinding or routing – Combined circuit switching and packet switching

Reexamination Certificate

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Details

C370S401000

Reexamination Certificate

active

06618368

ABSTRACT:

BACKGROUND OF THE INVENTION
The present invention relates to a multimedia communication system. More specifically, the present invention relates to an audio data gateway, an audio data relaying method, and a multiplexing format for the relaying of audio data.
There has been rapid growth of the Internet, which is based on TCP/IP. The WWW (World Wide Web), a global data communication system, has provided further acceleration to the widespread use of the Internet, and the transfer of multimedia data, including audio and image data, over the Internet has become commonplace.
Against this backdrop, communication systems referred to as Internet telephony are emerging to provide communication similar to telephones by connecting communication terminals to the Internet for sending and receiving real-time audio data. There have been increased efforts recently to unify communications over the Internet. In particular, there has been significant interest in Internet telephones where a phone call is made by using a standard telephone to access an access point on the Internet. The phone call goes from there to an access point that is near the person being called. From this access point, the call goes to the public telephone network. For example, if access points are set up at both Tokyo and Osaka, the phone charge for a call from Tokyo to Osaka would be made up of the phone charge from the user to the Tokyo access point and the phone charge from the Osaka access point to the person being called. This would be cheaper than using the public telephone network to make the call from Tokyo to Osaka. This type of service is expected to grow in the future. Internet telephony is presented in “Cheap calls using the Internet: Three minutes from Tokyo to Osaka for 55 yen” (Nikkei Communication, Feb. 3, 1997), and “Internet telephony and video phones” (PCWAVE, pg. 81, March 1997).
With the conventional communication systems described in the above references, if a single audio data gateway is used for multiple lines, gateway operations are performed by generating separate audio streams for each line. To send the audio data over the Internet, these communication systems must add to the audio data their own headers or headers as set forward by standards committees, as well as header information from the different communication protocol layers. Packets must then be formed from this data. The audio data itself is split up into very short packets, e.g., less than 100 ms. Thus, the size of the additional data becomes many times the size of the audio data. Multiple audio streams relayed to a particular target gateway will take up excessive bandwidth despite the fact that much of the added data in the streams is similar. As an example, we will look at packets based on the UDP/IP communication protocol widely used over the Internet and the RTP real-time communication protocol that has been proposed by the IETF (Internet Engineering Task Force), an industry standards association. In these packets, 30 ms of audio data (using G.723.1 encoding at 5.3 kbps) would take 20 bytes, the RTP header would take 2 bytes, the UDP header would take 8 bytes and the IP header would take 20 bytes. Thus, 40 bytes of header data is twice the size of the audio data. If different applications add additional header data to this, the bandwidth taken up by the header increases.
On the input side, analog audio data is converted to digital audio data, and digital audio data is converted to analog audio data on the output side. Since conversion between analog and digital data is performed in hardware, input/output operations between the hardware and the software must be performed, and this takes up CPU power. In general, input/output operations between hardware and software take up a few milliseconds. When generating packets at 20 ms intervals, input/output operations take place 50 times a second per line. Since each operation takes a few milliseconds, the processing time required for 50 input/output operations is at least 100 ms. If there are multiple active lines, the input/output operations will be more frequent. This is a significant factor in audio communications, which must be performed in real time.
An object of the present invention is to provide a data gateway and a data relaying method that can reduce the bandwidth used by multiple streams having the same destination by multiplexing data that can be shared such as header data.
Another object of the present invention is to provide a data gateway and a data relaying method that can reduce the number of input/output operations between hardware and software that require CPU power for streams that can be multiplexed.
Yet another object of the present invention is to provide a data gateway and a data relaying method where, if multiple streams have the same destination gateway and can be multiplexed, a gateway that handles multiple lines can negotiate with a destination gateway to determine whether to perform multiplexing and whether the destination gateway supports multiplexing.
Yet another object of the present invention is to provide a multimedia communication system that uses the audio data gateway described above.
Yet another object of the present invention is to provide a control unit that uses the audio data gateway described above and that performs administrative tasks for the audio data gateway device.
OBJECT AND SUMMARY OF THE INVENTION
In order to achieve the objects described above, the present invention provides an audio data gateway that includes: means for receiving an Internet relay request and a number called in a call from a telephone via the Internet, selecting a destination audio data gateway using the number called, and performing relay operations with the selected destination audio data gateway; and means for multiplexing relay audio data for streams based on multiple relay requests having the same relay destination, i.e. streams bound for the same destination and having common data so that the relay audio data in these streams (the non-shared data of these streams) are multiplexed into a single relay stream.
In this communication system, a plurality of relay streams are multiplexed into a single relay stream. This makes it possible to conserve more bandwidth compared to relaying each relay stream independently.
Also, the hardware for performing analog/digital conversion of audio data includes means for performing batch transfer to higher-level software of multiple sets of converted audio data that are bound for the same relay destination.
Since this communication system performs input and output operations between hardware and software all at once, the number of input and output operations, which require CPU power, can be reduced and more lines can be handled.
In the audio data gateway according to the present invention, the packet format for multiplexed streams includes at least an identifier to indicate whether the packet is a multiplexed stream, as well as means for negotiating with the destination on whether multiplexing is to be performed when a new stream is created. This audio data relaying system also includes a control unit for performing administrative operations on the audio data gateway. The control unit includes: means for monitoring, which monitors proper operation of the audio data gateways; relay logs (the phone numbers of users who made calls, the phone numbers called, the times at which calls were begun, the times at which calls were terminated, and the like) for each audio data gateway; means for periodically collecting relay packet loss and relay packet delays; and means for sending scheduling information for each audio data gateway.
In the data relaying method according to the present invention, a request for a call and a telephone number to call are received from a telephone. Using the number being called, the gateway closest to the destination is found and the address of the resulting destination gateway is obtained. A connection request is established using the address of this gateway. Analog audio data from the telephone is converted into dig

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