Multiplex communications – Data flow congestion prevention or control – Flow control of data transmission through a network
Reexamination Certificate
1999-12-16
2004-05-11
Ho, Duc (Department: 2665)
Multiplex communications
Data flow congestion prevention or control
Flow control of data transmission through a network
C370S352000, C370S410000, C370S465000
Reexamination Certificate
active
06735175
ABSTRACT:
TECHNICAL FIELD OF THE INVENTION
The present invention is related to telecommunications. More particularly, the present invention is related to methods for changing quality of service for Voice over IP communications.
BACKGROUND
Information technology (IT) and Communication Networks (Networks) provide end users with access to communications and diverse media. Typical Networks include, for example: PSTN (Public Switched Telephone Network), IP-based (Internet Protocol, or Internet), ISDN (Integrated Services Digital Network), and PLMN (Public Land Mobile Network) Networks provide end users with diverse applications such as: wired and wireless voice, multimedia access/transmission, Internet access, Voice over IP (Internet Protocol) etc. Access to and communication over Networks is accomplished via hardware (e.g, switches, routers, servers, gateways, gateway controllers, etc.), and communications protocols (e.g., IP (Internet Protocol), RLP (Radio Link Protocol), ATM (Asynchronous Transfer Mode), GSM (Global System for Mobile Communication), CDMA (Code Division Multiple Access), TDMA (Time Division Multiple Access), RTP ( ), SS7 (Signaling System 7), SIP (Session Initiation Protocol), MGCP (Media Gateway Control Protocol), etc.).
A Network link may generally be set up by using access technology, such as PRA (Primary Rate Access) signaling or ISUP (ISDN user part). Functionality to set up links between, for example, a Media Gateway (MG) and the Internet, normally reside on a separate hardware component such as an Access Server (AS). Once the AS gets signaled by the Mobile Switching Center (MSC), an incoming communication requesting access is directed from the MSC towards the Media Gateway Controller (MGC). Once the MGC gets signaled via a communication requesting Internet access, the MGC seizes AS functionality.
Internet applications such as IA (Internet Access), VoIP (Voice over IP) and Multimedia generally require certain functionality such as contained within a MGC to control the media stream in the MG as well as other Internet specific functionality such as AAA (Authentication, Authorization, Accounting).
Networks access and operational inefficiencies generally occur because of bandwidth limitations or hardware clashes. For example, a traditional telephone call is established over a dedicated network circuit (e.g., Public Service Telephone Network, or “PSTN”). Regardless of the amount of voice data (e.g., analog voice signals) to be transferred across the circuit connection (e.g., via the PSTN), the entire circuit must generally be dedicated for the duration of the call; thus resulting in a potentially low overall usage of switching resources. An alternative to circuit switched (voice) calls is packet switched calls (e.g., voice over IP). In a packet switched call, the voice data is encoded into packets which are then directed across a network and reassembled into an approximation of the original data at the receiving end. During a silent period of a call, no data generally needs to be packetized and transmitted across the network; thus saving network resources and allowing for higher call density (e.g., more calls can be routed on the same amount of network resources).
In order for a call to be sent over a packet-switched network, the voice data (e.g., analog voice signals) must be sampled and converted into a digital format (e.g., packetized voice). The sampling process introduces a trade-off between sound quality and bandwidth utilization. A voice stream can generally be sampled very frequently to produce very high sound quality on the receiving end, but at the cost of using a large amount of bandwidth to transport generated packets. Alternatively, voice can be sampled infrequently to produce a smaller amount of packet data but at the cost of low sound quality at the receiving end.
As part of setting up a call over a packet network, a codec (coding/decoding algorithm) must generally be specified to control what rate of sampling is used to encode the call. Depending on the codec algorithm specified, various levels of voice quality/bandwidth utilization can be generally achieved.
There is apparently a lack of available methods and systems for a caller making a call on a voice over IP network to specify, during a call, a level of voice quality or Quality of Service (QoS) that is desired. While the network operator can generally specify at initial call setup or even in mid-call what quality level will be used, the caller apparently cannot. Such a lack of user control over QoS precludes the network operator from offering low Quality of Service (QoS) at a low cost while still allowing the caller to upgrade to a higher QoS at a higher cost as needed during, or for the remaining duration of, a voice communication.
SUMMARY OF THE INVENTION
Aspects of the present invention provide methods and/or systems whereby a subscriber making a call on a Voice over IP network can choose a QoS during an ongoing voice communication, and may be realized over numerous network configurations so long as the configuration provides for: packetization of voice streams over an IP network using a codec; codec choices; the ability to change a codec in use during an ongoing call; and the ability for the caller to signal the network (e.g., through in-band DTMF (dual tone multifrequncy) commands via digit collection and/or through a flash feature) when a change of codec (coder/decoder) algorithm is desired.
One aspect of the invention provides a method for changing QoS for voice over IP communication wherein caller invoked signaling of a network controller indicates that a change in QoS is desired during an ongoing voice over IP communication; and system implementation of a change in QoS is provided through packetization or depacketization of a communication in response to the caller invoked signaling.
Another aspect of the present invention provides a system for changing QoS for voice over IP communications. A signal monitoring module for monitoring subscriber inputs representing requests for a QoS change; and a controller for implementing subscriber inputs representing requests for a QoS change.
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SIP Protocol Specification (IETF RFC 2543).
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