Electrical computers and digital processing systems: multicomput – Computer-to-computer protocol implementing – Computer-to-computer data streaming
Reexamination Certificate
1999-02-02
2002-03-19
Lim, Krisna (Department: 2757)
Electrical computers and digital processing systems: multicomput
Computer-to-computer protocol implementing
Computer-to-computer data streaming
C370S516000, C710S052000, C713S401000
Reexamination Certificate
active
06360271
ABSTRACT:
RELATED APPLICATIONS
The present document is related to two other U.S. patent applications filed on the same date, each in the name of the same inventors. The other two applications are entitled “System for Adjusting Billing for Real-Time Media Transmissions Based on Delay” and “System for Routing Real-Time Media Transmissions Based on Delay.” The entirety of each of these other applications is hereby incorporated herein by reference.
BACKGROUND OF THE INVENTION
1. Field of the Invention
The present invention relates to the transmission of real-time media signals over data networks and more particularly to a method and apparatus for using absolute time information and/or other parameters to assess, improve and manage such transmission. The invention is particularly useful in the context of IP networks such as the Internet or an intranet. However, the invention is not limited to use in this context but extends more generally to use in the context of any store-and-forward network such as any packet switched network, including, for instance, ATM, frame relay, X.25 and SNA networks.
2. Description of Related Art
There has long been a need in the art to transmit real-time media signals from one location to another. In early days, the need to convey voice signals was satisfied through the use of relatively simple analog telephone systems. More recently, the availability of digital telephone systems and advanced computer networks such as the Internet has facilitated the communication of assorted real-time media signals, such as voice, audio and/or video over long distances at a fraction of the cost of conventional systems. Currently, there are two types of networks that can be used to convey real-time media signals, circuit switched networks and packet switched networks.
In a circuit switched network, a point-to-point communication path or circuit is established between two or more users, such that the users have exclusive and full use of the circuit until the connection is released. A media signal to be transmitted is then sent in whole over the dedicated circuit, received by the other side and played out to a user. The public switched telephone network is an example of a circuit switched network.
In a packet switched network, in contrast, a message to be sent is divided into blocks, or data packets, of fixed or variable length. The packets are then sent individually over the network through multiple locations, and then reassembled at a final location before being delivered to a user at a receiving end. To ensure proper transmission and re-assembly of the blocks of data at the receiving end, various control data, such as sequence and verification information, may be appended to each packet in the form of a packet header, or otherwise associated with the packet. At the receiving end, the packets are then reassembled and transmitted to an end user in a format compatible with the user's equipment. The Internet is an example of a packet switched network.
At their inception, each type of telecommunications network was designed to support the transmission of select types of media. Circuit switched networks were designed to carry real-time audio signals (e.g., voice). Packet switched networks, on the other hand, were designed to carry pure data signals (e.g., e-mail). Today, however, these networks compete to provide multi-media transmission services, including, for instance, the transmission of data, voice, audio and/or video. Further, with the growth of the Internet and other advances in technology, packet switched networks are now competing with conventional circuit switched networks to provide interactive communications services such as telephony and multi-media conferencing. In the context of packet switched networks operating according to Internet Protocol (IP), this technology is presently known as internet telephony, IP telephony or, where voice is involved, Voice over IP (VoIP).
Internet telephony presents an attractive technology for use in long distance telephone calls, as compared to the public switched telephone network (PSTN), which has been the traditional transmission medium. One of the primary advantages of internet telephony is its flexibility and features, such as the ability to selectively provide different levels of service quality and to integrate voice and data services (for instance, integrating e-mail and voice mail functions).
Another primary advantage of internet telephony is cost. In the United States, for instance, long distance service providers for the PSTN provide domestic services at rates ranging from roughly 10 to 30 cents per minute, and international rates for substantially more, depending on the time of day, day of the week, and the distances involved. In contrast, the cost of an internet telephony call anywhere in the world is potentially the cost of a local telephone call to a local internet telephony service provider at one end and the cost of a local call from an internet telephony service provider at the far end to the destination telephone. Once the call is routed from the local internet telephony provider onto the IP network, the cost to transmit the data from the local internet telephony provider to the far end internet telephony provider can be free for all practical purposes, regardless of where the two parties are located. Similarly, the cost to facilitate a direct dial internet telephony call can theoretically be free, except for possible access fees charged by local exchange carriers. Internet telephony service providers can thus potentially charge users far less for internet telephony calls than the users would pay for comparable calls placed strictly over the PSTN.
To transmit a real-time media signal over a packet switched network, the media signal is typically first sampled, divided into frames, and channel coded or compressed according to an established media coding standard. Each encoded frame of data is then inserted as payload into a packet, which is then labeled with one or more headers (often depending on various transmission protocols). The header usually identifies a packet sequence number, a source and destination network addresses for the packet, and a sender timestamp.
In general, a purpose of the sender timestamp is to record the time spacing between packets in a sequence. Therefore, the sender timestamp may identify any suitable time at the transmitting end, consistently for the packets in a sequence. For instance, without limitation, the sender timestamp may identify when the first sample of the payload in a packet was taken or when the packet was sent into the network.
In this regard, each packet of a real-time media sequence typically represents a successive time block of the underlying media signal. For instance, according to the G.723.1 standard, a 16 bit PCM representation of an original analog speech signal is partitioned into consecutive segments of 30 ms length, and each of these segments is encoded into a frame of 240 samples, represented by either 20 or 24 bytes (depending on a selected transmission rate). The time spacing between each of these frames is significant, as it serves in part to define the underlying signal. For example, under G.723. 1, it is important to know that a sequence of four packets were transmitted at times t, t+30, t+60, and t+90. With this inter-packet time spacing information and sequence number information, a receiving device ideally should be able to reconstruct the packet sequence and decode and play out the underlying signal.
As a stream of real-time media packets is created, each packet is sent independently into the network and routed to the receiving end as identified by the destination address in the packet header. The packets may be sent back to back or with a holding time between packets. Ideally (excepting packet loss, for instance), each packet will then traverse the network and arrive at the destination end, to be decoded and played out to an end user.
As is well known in the art, the transmission of any data signal from one location to another over a te
Borella Michael S.
Grabiec Jacek A.
Schuster Guido M.
Sidhu Ikhlaq S.
3Com Corporation
Lim Krisna
McDonnell & Boehnen Hulbert & Berghoff
LandOfFree
System for dynamic jitter buffer management based on... does not yet have a rating. At this time, there are no reviews or comments for this patent.
If you have personal experience with System for dynamic jitter buffer management based on..., we encourage you to share that experience with our LandOfFree.com community. Your opinion is very important and System for dynamic jitter buffer management based on... will most certainly appreciate the feedback.
Profile ID: LFUS-PAI-O-2834008