Telecommunications – Transmitter and receiver at same station – Radiotelephone equipment detail
Reexamination Certificate
2000-05-17
2004-06-15
Chin, Vivian (Department: 2682)
Telecommunications
Transmitter and receiver at same station
Radiotelephone equipment detail
C455S422100, C370S468000, C370S352000, C370S401000
Reexamination Certificate
active
06751477
ABSTRACT:
TECHNICAL FIELD OF THE INVENTION
The present invention is directed, in general, to wireless communications systems and, more specifically, to systems and methods for dynamically optimizing the fidelity of a speech signal received from a wireless telephony device and transmitted through a packet-switched network.
BACKGROUND OF THE INVENTION
The world is currently experiencing revolutionary changes in communication systems, brought about in part by the general availability of access to the Internet. In particular, interest in Internet Protocol (IP) telephony, or Voice over IP (VoIP), has expanded rapidly as the associated technologies have matured. The interoperability of IP networks with other voice communications networks, such as the Public Switched Telephone Network (PSTN) and wireless communications networks, however, is a prominent factor in the eventual success of Internet telephony.
The telecommunications industry has also recently undergone a revolution in the area of wireless telephony; e.g., “cellular” or “mobile” devices. Wireless telecommunications networks are an adjunct to the PSTN, and depend upon the circuit-switched network of the PSTN for communications with conventional wired telephony devices as well as with mobile devices. Thus, the routing of information to a mobile device communicating with a wireless telecommunications network has heretofore also required such information to be routed through the PSTN.
Because of the proliferation of both IP networks and wireless telephony systems, it is desirable to develop systems and methods to route voice calls from wireless telephony devices over IP networks. One of the deficiencies of IP networks, however, is that such networks are not inherently suited to the transport of voice information. This is due to the fact that such networks can suffer from delays in delivering packets and limitations in bandwidth, which are factors related to the QoS of the IP network, thus affecting the “fidelity” of the received voice information. Furthermore, the QoS of the IP network may vary over the duration of a voice call. Thus, there is a need in the art for systems and methods for optimizing the fidelity of a speech signal received from a wireless telephony device and transmitted through a packet-switched network, such as an IP network. In particular, there is a need in the art for systems and methods for continuously, or “dynamically,” optimizing the fidelity of such speech signals over the duration of a voice call.
SUMMARY OF THE INVENTION
To address the above-discussed deficiencies of the prior art, the present invention relates to systems and methods for dynamically optimizing the fidelity of a speech signal received from a wireless telephony device and transmitted through a packet-switched network. In an exemplary embodiment described hereinafter, at least one Quality of Speech (QoS) transmission characteristic is determined for the packet-switched network intermediate to the a first Media Gateway (MGW) and a second MGW coupled to the packet-switched network; the fidelity of speech transmitted through the packet-switched network being a function of the QoS transmission characteristic. A speech codec is dynamically-selected as a function of the QoS transmission characteristic of the packet-switched network. The speech signal is encoded into speech data using the dynamically-selected speech codec, and transmitted through the packet-switched network from the first MGW to the second MGW. The second MGW decodes the speech data using the dynamically-selected speech codec.
The identification of the dynamically-selected speech codec can also be communicated to the wireless telephony device to allow it to adapt its encoding format to a preferred format for a given QoS of the packet-switched network. The QoS transmission characteristic of the packet-switched network can be, for example, an average packet-delay or packet throughput. In addition, the dynamically-selected speech codec can be further selected as a function of an air-interface Quality Indicator (QI) provided by the wireless telephony device. In such embodiments, the fidelity of the received speech signal can be optimized as function of the quality of both the wireless transmission path and the packet-switched network (which can comprise or include wireless transmission paths). In implementations in which the wireless communications network conforms to the Global Standard for Mobile Communications (GSM), the dynamically-selected speech codec can be Enhanced Full Rate (EFR), Full Rate (FR), and Half Rate (HR).
In an implementation particularly described hereinafter, the speech data is transmitted through a packet-switched network using an Internet Protocol (IP) based network layer; the scope of the invention, however, includes the use of other packet-switched protocols, such as Asynchronous Transfer Mode (ATM). In addition to the use of IP, the speech data can be transmitted using a transport layer protocol such as User Datagram Protocol (UDP) or Transport Control Protocol (TCP).
Upon receipt of speech data at the destination MGW, it may be necessary to encode the data into a different format for transmission through another network. For example, if the destination of the speech data is a telephony device coupled to the Public Switched Telephone Network (PSTN), the destination MGW, in addition to decoding the speech data received over the packet-switched network using the dynamically-selected speech codec, must further convert the speech data to Pulse Code Modulated (PCM) data for transmission over the PSTN.
The foregoing has outlined, rather broadly, the principles of the present invention so that those skilled in the art may better understand the detailed description of the exemplary embodiments that follow. Those skilled in the art should appreciate that they can readily use the disclosed conception and exemplary embodiments as a basis for designing or modifying other structures and methods for carrying out the same purposes of the present invention. Those skilled in the art should also realize that such equivalent constructions do not depart from the spirit and scope of the invention in its broadest form, as defined by the claims provided hereinafter.
REFERENCES:
patent: 6192395 (2001-02-01), Lerner et al.
patent: 6256487 (2001-07-01), Bruhn
patent: 6445697 (2002-09-01), Fenton
patent: 6600738 (2003-07-01), Alperovich et al.
patent: 6603774 (2003-08-01), Knappe et al.
patent: 6611694 (2003-08-01), Oltedal et al.
patent: 6633582 (2003-10-01), Panburana et al.
patent: 2001/0043577 (2001-11-01), Barany et al.
patent: 0 966 145 (1999-12-01), None
patent: WO 99 05830 (1999-02-01), None
patent: WO 99 53700 (1999-10-01), None
Bruhn S et al: “Concepts and solutions for link adaptation and inband signaling for the GSM AMR speech coding standard” Vehicular Technology Conference, 1999 IEEE 49th Houston, TX, USA May 16-20, 1999, Piscataway, NJ, USA, IEEE, US May 16, 1999, pp. 2451-2455, XP010342317 ISBN: 0-7803-5565-2 p. 2451, column 2, line 5-line 10 and page 2451, column 2, line 15-line 18.
Alperovich Vladimir
Timberlake Andreea
Chin Vivian
Persino Raymond B.
LandOfFree
Systems and methods for dynamically optimizing the fidelity... does not yet have a rating. At this time, there are no reviews or comments for this patent.
If you have personal experience with Systems and methods for dynamically optimizing the fidelity..., we encourage you to share that experience with our LandOfFree.com community. Your opinion is very important and Systems and methods for dynamically optimizing the fidelity... will most certainly appreciate the feedback.
Profile ID: LFUS-PAI-O-3295804