The present invention relates generally to packet switched communications networks and, in particular, to a system providing efficient transport of coded data over a packet-based core network.
The usage of, and demand for, mobile telecommunications continue to increase at a staggering rate. Wireless telecommunications service providers are constantly seeking new ways to improve and expand the services they provide while lowering their investment and operational costs. This ever-increasing demand has driven the development of new and improved topologies and protocols for wireless communications systems. It is now possible to route voice communications, in packetized form, over Internet Protocol (IP) systems conventionally associated with computer data communications. Such capabilities hold the promise increasing efficiency and decreasing costs associated with wireless communications.
Interest grows in IP-based communications as an alternative to conventional circuit switched systems. Circuit switched systems require dedicated channels, reserving an ISUP (ISDN user part) link for any given communication. Therefore, any given call effectively monopolizes a line (e.g. trunk or E1/T1 line) between call origin and destination; requiring a separate line for each call processed. Even in conventional xe2x80x9cwirelessxe2x80x9d communications systems, a call is generally only wireless between the mobile unit and its closest base station, which thereafter typically routes the call on circuit switched infrastructure. For example, in a typical GSM (Global System for Mobile communications) network, once a signal is received at the base station, it is thereafter routed via circuit switched infrastructure to the mobile switching center (MSC) and the rest of the GSM system.
It should thus easily be appreciated that as demand continues to increase, infrastructure associated with circuit switched systems must increase correspondingly. This results in increased system overhead, reduced call volume bandwidth, and increased user costs to cover the additional overhead.
In comparison, IP communications packetize voice data for transmission over existing IP networks; enabling users to communicate (e.g. via phone calls or computer-based conferencing applications) as long as they want for only the cost of the access to the IP network. IP infrastructure is ubiquitous; and use of IP infrastructure is not dedicated (i.e. multiple users utilize, one packet at a time, the same resources), lowering system overhead and use costs.
Although IP network communication is, in some respects, advantageous over circuit switched communication, other considerations limit the commercial usefulness of conventional IP network implementations. Consider, for example, a wireless communications scenario where communication originates in a radio access network (e.g., universal mobile telephony system (UMTS)), is transferred across an IP based core network (backbone), and is delivered to an external, circuit-switched network (e.g., public switched telephone network (PSTN)). Generally, radio access networks typically utilize low bit rate speech encoding (e.g., 13 Kb/s), while traditional circuit switched networks utilize high bit rate speech encoding (e.g., 64 Kb/s). Conversion and formatting from one encoding to the other is required to successfully deliver speech data from one network to the other. The conversion and formatting functions are usually executed by a CODEC (Coder-DECoder).
Conventional communications systems generally perform the conversion from lower bit rate encoding to higher bit rate encoding as the data enters the network backbone. As data from the radio access network enters a media gateway (MGW) that serves as its interface with the IP backbone, the conversion is executed by a CODEC in the media gateway. The data is then transferred across the IP network at the higher bit rate encoding, delivered to another MGW for transfer to the external public network. Thus, the higher bit rate encoded speech must be packetized for transmission over the IP backbone. This is less efficient than packetizing and transmitting the lower bit rate encoded speech, and decreases effective system bandwidth. In many conventional systems, even where the call will terminate in a similar radio access network, speech data is superfluously converted twice: from low to high bit rate upon entering the IP backbone, transmitted over the IP backbone at high bit rate, and from high to low bit rate upon transfer out of the IP backbone. This is obviously inefficient, and degrades overall system performance.
Some previous systems have attempted to overcome these limitations by eliminating the conversion to the higher bit rate altogether. This is typically achieved by negotiation of the CODEC in the terminating MGW (i.e. the one receiving the call) with the CODEC in the originating MGW, whereby each CODEC essentially performs no conversion. Speech data is thus introduced to, transmitted over, and delivered from the IP backbone at the lower bit rate encoding. Although this approach is, in certain ways, advantageous over prior solutions, it is still impractical as it Liz fails in many common call flow scenarios, such as call forwarding to a PSTN. This results in reduced system reliability or, alternatively, system inefficiency due to redundant call processing and constructs.
From the foregoing, it can be appreciated that a need exists for a providing efficient, reliable, and cost-effective IP network communication in a variety of wireless telecommunication applications. It is desirable that such a system provide robust and versatile structure and methods by which low bit rate encoded data may be transported over a packet based network, even if the destination for the data is a high bit rate external network; overcoming the limitations of conventional systems.
The present invention provides a system for low bit rate encoded data transport over a packet based network, regardless of whether the destination for the data is a low or high bit rate network. The present invention provides a dual CODEC functionality by which conversion to high bit rate encoding is executed after transport across a packet based network, and only when required by the terminating network.
More specifically, the present invention provides a packet-based telecommunication system comprising a first media gateway having a first CODEC structure and a second media gateway having a dual function CODEC structure, wherein the dual function CODEC structure is adapted to provide tandem free operation between itself and the first CODEC structure.
The present invention further provides a dual function CODEC, utilized in packet based telecommunications, comprising a first element adapted to negotiate tandem free operation, and a second element communicatively coupled to the first element and adapted to selectively convert data coding responsive to the first element.
The present invention also provides a method of providing efficient communication between a low bit rate network and a high bit rate network over a packet based network by providing within the packet based network a first CODEC associated with the low bit rate network, providing within the packet based network a second CODEC associated with the high bit rate network, transferring low bit rate data from the first CODEC to the second CODEC, and using the second CODEC to convert the low bit rate data received from the first CODEC to high bit rate data.