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COMMUNICATING VOICE OVER A
RELATED APPLICATIONS AND CLAIM OF
This application is a continuation of and claims priority to U.S. patent application Ser. No. 10/409,994, filed on Apr. 8, 2003, now U.S. Pat. No. 6,768,733 entitled "COMMUNICATING VOICE OVER A PACKET-SWITCHING NETWORK", which is a continuation of and claims priority to U.S. patent application Ser. No. 09/163,312, now issued as U.S. Pat. No. 6,570,869, filed on Sep. 30, 1998, entitled "COMMUNICATING VOICE OVER A PACKETSWITCHING NETWORK", the contents both of which are hereby incorporated by reference in their entirety for all purposes.
FIELD OF THE INVENTION
The present invention relates to telecommunications and more particularly to packet switched networking systems capable of carrying voice traffic.
BACKGROUND OF THE INVENTION
Recent legislative changes in the United States have promoted competition in the telecommunication industry and spurred demand for new services at lower prices. These trends are pressuring major telecommunications carriers to increase capacity while reducing the cost of providing service. Consequently, major carriers around the world are looking to packet technologies, such as Internet Protocol (IP), frame relay, and Asynchronous Transfer Mode (ATM), to replace circuit-switched technologies in the Public Switched Telephone Network (PSTN) for providing voice capability. In addition, IP, frame relay, ATM, and other packet-based technologies offer narrow-band and broadband services to selected customers on the same network, providing the same platform for integrated voice, data, and video services from low bandwidth to very high bandwidths.
Over the decades, however, major voice carriers have invested heavily in developing a Signaling System 7 (SS7) signaling and switching infrastructure to offer reliable telephone service. This infrastructure includes countless systems for billing, provisioning, maintenance, and databases that are compatible only with SS7. These systems are commonly referred to "legacy systems," a term that also includes other proprietary protocols such as ISDN_PRI, DPNSS, ISUP, TUP, NUP, H.323, and SIP. Due to the substantial investment in the legacy systems, it is desirable to keep the legacy systems in operation, yet still take advantage of the newer packet technologies.
These legacy systems, however, do not handle the protocols for the newer packet-switching networks, and, due to the age of many of the legacy systems, it is difficult and expensive to upgrade or replace the legacy systems to support the newer packet-switching protocols.
Accordingly, there exists a need for establishing and carrying voice calls that are originated or terminated by legacy systems over a packet-switching network. There is also a need for a way to seamlessly integrate legacy SS7type systems and newer packet-switching networks.
Moreover, certain demographic trends are motivating telephone call carriers to integrate their systems with packetswitched networks. Certain countries are known to generate an above-average amount of long-distance telephone traffic.
For example, residents of Israel are known to consume long-distance telephone services at a rate far greater than the average of residents in other industrialized nations. Longdistance telephone services carried over the PSTN are
5 expensive. Voice calls carried over the globally accessible packet-switched network known as the Internet, however, are generally free. Accordingly, local telephone companies and other call access providers in certain countries are acutely interested in finding ways to integrate the PSTN with
10 the Internet.
BRIEF DESCRIPTION OF THE DRAWINGS
The present invention is illustrated by way of example, 15 and not by way of limitation, in the figures of the accompanying drawings and in which like reference numerals refer to similar elements and in which:
FIG. 1 is a diagram of a packet-switching network carrying voice signals; 20 FIG. 2 is a block diagram of a signaling unit;
FIG. 3 is a block diagram of a software architecture of a signaling unit;
FIG. 4 is a call flow diagram illustrating an establishment of a voice call and voice call release over a packet-switching 25 network;
FIG. 5(a) is a diagram of another packet-switching network carrying voice signals;
FIG. 5(b) is a diagram of still another packet-switching network carrying voice signals; and 30 FIG. 5(c) is a diagram of yet another packet-switching network carrying voice signals.
DESCRIPTION OF THE PREFERRED
A telecommunications method, network, and devices for carrying voice over a packet-switching network are described. In the following description, for the purposes of explanation, numerous specific details are set forth in order
40 to provide a thorough understanding of the present invention. It will be apparent, however, to one skilled in the art that the present invention may be practiced without these specific details. In other instances, well-known structures and devices are shown in block diagram form in order to
45 avoid unnecessarily obscuring the present invention.
FIG. 1 depicts a telecommunications network that carries
50 voice calls from an originating node 100 to a terminating node 160 over a packet-switching network 130, in which the voice signaling processing is separated from the processing of the voice data. More specifically, the voice signaling aspects of establishing and handling voice calls over packet
55 switching network 130 are provided by one or more signaling units, for example, the originating signaling unit 120 and the terminating signaling unit 140. The aspects relating to the voice traffic of a voice call are handled by one or more coding units, for example, the originating coding unit 110
60 and the terminating coding unit 150.
For purposes of illustration, FIG. 1 depicts a network configuration in which the originating coding 110 and the terminating coding unit 150 are coupled to respective signaling units, namely originating signaling unit 120 and
65 terminating signaling unit 140. As described in more detail hereinafter, however, the signaling processing functionality for the originating signaling unit 120 and the terminating
signaling unit 140 can be incorporated into a single signaling unit. Even though the signaling units and the coding units are generally described herein in terms of being separate devices, which can be geographically remote from one another, a signaling unit and a coding unit may also be 5 incorporated as respective subsystems of a single computer system. Thus, the present invention is not limited to the configuration depicted in FIG. 1.
Originating node 100 can be implemented as a Private Branch eXchange (PBX), a telephone switch, a "smart 10 phone" capable of generating voice calls, a wireless PBX, or a legacy telecommunications system. Similarly, terminating node 160 can also be a PBX, telephone switch, telephone, a wireless PBX, or legacy telecommunications system.
Packet-switching network 130 is a network designed to carry information in the form of digital data packets. In such a network, data to be transmitted is subdivided into one or more individual packets of data, each having a unique identifier and a destination address. Each packet is individu- 2Q ally routed or switched to the destination address, and individual packets for a single body of data may traverse the packet-switching network by different routes. In fact, the individual packets may even arrive at the destination in a different order from which they were shipped, to be reas- 25 sembled at the destination in the proper sequence based on the packet identifiers. Packet-switching network 130 can be implemented as an IP network, an ATM network, a frame relay network, or by any other packet-switching technology. In some implementations, the packet-switching network 130 3Q may even be overlaid on the PSTN. One example of packet-switching network 130 is the global packet-switching network known as the Internet.
The telecommunication network of FIG. 1 includes an originating coding unit 110 and a terminating coding unit 35 150 functioning as gateways between the respective originating node 100 and the terminating node 160 and the packet-switching network 130. The originating coding unit 110, coupled to the originating node 100 by a trunk such as a Tl line or an El line, converts multiplexed voice data 40 produced by originating node 100 into packets for the packet-switching network 130. The voice data produced by originating node 100 may be, for example, Time Division Multiple Access (TDMA) and Code Division Multiple Access (CDMA) information. The originating coding unit 45 110 can also be configured to support voice data encoding and decoding as well as associated functions such as echo cancellation, voice activity detection, and voice compression. Similarly, the terminating coding unit 150 is also configured to convert between multiplexed voice data and 50 voice data packets as well as the encoding and decoding functions.
While a major purpose of the origination coding unit 110 is to terminate the bearers from PBX 100, in some embodiments the originating coding unit 110 is also configured to 55 extract or "groom" the signaling data associated with the incoming voice call from originating node 100. This signaling data is then transmitted or "backhauled" over a backhaul signaling link 112 to a signaling apparatus such as originating signaling unit 120. The backhaul signaling link 112 can 60 be implemented in various ways, including by an IP connection over Ethernet or other Local Area Network (LAN) technology such as token ring. The signaling data in the voice call can be Channel Associated Signaling (CAS), in which the signaling bits are isolated, time stamped, pack- 65 aged in IP or ATM packets, and shipped to the originating signaling unit 120.
Similarly, the terminating coding unit 150 is coupled by a backhaul signaling link 152 to a signaling apparatus such as terminating signaling unit 140. The terminating coding unit 150 is configured for receiving signaling messages from the terminating signaling unit 140 and appropriately transmitting them to the terminating node 160. Preferably, the coding units are implemented to be symmetrical, supporting the functionality of both the originating coding unit 110 and the terminating coding unit 150 as described herein. In fact, a single coding unit can performing the both the originating and terminating functionality for the same call.
Alternatively, the signaling data can be Common Channel Signaling (CCS), such as an ISDN PRI, in which case the signaling data is directly transported to the originating signaling unit 120. In an embodiment wherein originating node 100 implements a CCS signaling such as U.S. SS7 signaling, the signaling data can be directly transmitted over link 113 to the originating signaling unit 120 bypassing the originating coding unit 110 entirely. Similarly, when terminating node 160 implements such signaling, the signaling data can be directly transmitted over link 153 from the terminating signaling unit 140 to the terminating node 160, bypassing the terminating coding unit 150. By these techniques, the originating signaling apparatus 120 is advantageously capable of receiving the signaling data associated with the voice call in a flexible manner, suitable for interfacing with diverse legacy systems.
The originating signaling unit 120 and the terminating signaling unit 140 implement a "virtual switch" and are responsible for processing and routing the signaling messages that are exchanged to set up and tear down a voice connection. Thus, the signaling units perform such functions as call resolution, call routing, bearer selection, and generation of call detail records (CDRs) for billing management. In one embodiment, the signaling units also convert the legacy protocols of the originating node 100 and the terminating node 160, such as DPNSS, ISDN_PRI, SS7/C7 (including ISUPs, TUPs, and NUPs), H.323, SIP, or CAS, into messages for communicating with one another and for controlling a coding unit over control links 114 and 154. Control links 114 and 154 can be implemented over IP or ATM and, in fact, on the same channel as the respective backhaul signaling link 112 and 152, respectively. Through the control link, a coding unit is controlled by a signaling unit, for example, to establish a bearer channel for the voice data over the packet-switching network 130.
In the configuration depicted in FIG. 1, a voice call from originating node 100 is received by the originating coding unit 110, which, if necessary, extracts the signaling data associated with the voice call and transmits the signaling data over the backhaul signaling link 112 to originating signaling unit 120. In response, the originating signaling unit 120 obtains the network address of the originating coding unit 110 within the packet-switching network 130 by accessing configuration data stored on the originating signaling unit 120, by querying the originating coding unit 110 over the control link 114, or by inquiring another computer system (not shown) such as domain name server (DNS).
Next, the originating signaling unit 120 determines which terminating signaling unit 140 should receive the call by accessing internal routing tables or querying external systems. After the originating signaling unit 120 has performed this call routing capability, the originating signaling unit 120 transmits a signaling message, including information for establishing the voice and the network address of the originating coding unit 110, through network 132 to terminating signaling unit 140.