Internet protocol based telephone system in which telephone traffic is transmitted at least in part over an Internet protocol based network retaining terminal telephones and exchanges

(54) (56) (57)

INVENTOR INVENTOR

Telefonaktiebolaget LM Ericsson, 126 25 Stockholm SE Jan Sjölund, Saltsjö-Boo SE, Ros-Marie Furtenback, Johanneshov SE, Jan Gjärdman, Farsta SE, Tomas Nyländer, Stavsnäs SE

OMBUD Albihns Stockholm AB

NAME Telecommunication system, access link and procedure based on IP

CALLED PUBLICATIONS:

WO Al 9 733 412 (H04L 12/56), WO Al 9 844 713 (H04M 7/00), WO A2 9 859 469 (H04L 12/56),

SUMMARY:

The invention relates to a hybrid arrangement which allows an existing wired telephone network to be utilized on a data packet switched network. The wired connected transport network has been replaced by an IP-based data packet switched network (30), while the wired infrastructure such as terminals, telephone exchanges and the like have been maintained. In a preferred embodiment of the invention, both a subscriber (10, 11, 20, 21) and a local telephone exchange (40, 50) are connected to, and separated by, an IP-based data packet switched network, such as the Internet (30). Signaling between the subscriber (10, 11,20,21) and the switch (40.50) is effected using a standard user / network protocol such as V5.1 for PSTN and ISDN BRI or DSS1 for ISDN PRI. This standard user / network protocol is superimposed on an IP-based network protocol, such as TCP or UDP on IP.

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ΓMationskoder

<img file="SE516306C2_D0002.tif" />

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516 306

TECHNICAL FIELD

The invention is directed to telephone systems. The invention has particular relevance in telecommunications systems where teletraffic is transmitted at least in part via an IP-based network, such as a data network or the Internet.

BACKGROUND OF THE ART

The benefits of transmitting voice information over a data packet or cellular network have long been realized. The relatively low cost of utilizing packet switched networks, such as the internet, instead of a wired network has generated growing interest and many telecom operators now claim that packet switching is about wired voice transmission in terms of bandwidth utilization in their networks. This is partly due to the increase in products that handle speech via IP functions. However, it has also become increasingly interesting for network operators to enable telephone calls originating in a standard wired network to be routed at least partially through a data packet switched network without any change in the way a user uses a telephone or other telecommunications equipment.

An example of such an arrangement is described in GB-A-2 331 197.1 In this known arrangement, a wired interurban network has been replaced with a data packet switched IP network. This has the advantage that the telecom operator is allowed to keep the costs of interurban calls down while the regular wired systems are still used.

However, while data packet switched technology is increasing in importance compared to wiring, there is a need to incorporate in a telecommunication system more of the advantages of packet switched features. However, if telecom operators continue to receive dividends on the substantial investment represented by a link-connected infrastructure, there is still a need to maintain as much of the line-connected system as possible.

THE INVENTION IN BRIEF

The invention provides a hybrid arrangement which allows an existing wired telephone network to be utilized for a data packet switched network. In essence, the underlying wired transport network is replaced with that of an IP-based data packet switched network, while the wired infrastructure such as terminals, telephone exchanges and the like are retained with modifications. In preferred embodiments of the invention, both a subscriber and a local telephone exchange are connected to, and separated by, a data network. The signaling between the subscriber and the switch is performed using standard user / network protocols, such as V5.1 for PSTN and ISDN Basic Frequency Interface (BRI) or DSS1 for ISDN Primary Frequency Interface (PRI). This standard user / network protocol is superimposed on an IP-based network protocol, such as TCP or UDP. By utilizing protocols conventionally used by the corresponding circuit switching system elements, the transition from wired to IP-based data packet switched network transport can be achieved in a fast and secure way as the network elements are interchangeable with different suppliers' equipment. Furthermore, the existing services, supported by the standard protocols, will be supported in the data packet switched network.

In addition, the connection of the subscriber, or the access node to which a subscriber is connected, to the IP-based network enables the full future utilization of the IP-based network without extensive modification of the system.

BRIEF DESCRIPTION OF THE DRAWINGS

Further objects and advantages of the present invention will become apparent from the following description of preferred embodiments given by way of example with reference to the accompanying drawings, in which

516 Fig. 306 is a schematic representation of a telecommunications network according to the present invention; Fig. 2 shows an arrangement for a telephone server; Fig. 3 shows an access node according to the present invention; and Fig. 4 shows the signaling between the various elements of the network of Fig. 1. Fig. 5 shows the interaction between the network of Fig. 1 and an adjacent data packet switched network; Fig. 6 shows a first embodiment of a network element adapted to interact with neighboring networks; 7 shows a second embodiment of a network element adapted to interact with neighboring networks and Figure 8 shows a third embodiment of a network element adapted to interact with neighboring networks.

DETAILED DESCRIPTION OF THE DRAWINGS

Fig. 1 shows a hybrid network of the present invention, where a call is in progress between an A subscriber 10 represented by the telephone A on the left of the figure, and a B subscriber 20 shown on the right of the figure. The subscribers are each connected to an access node 11,21 and these access nodes 11,21 are in turn connected to a data or similar IP-based data packet switched network 30, which is typically the Internet. Also connected to the data network 30 is a telephony server 40, which is the host server for the subscriber A 10 and controls the connection of calls to and from the subscriber A 10, and a similar host telephone server 50 for the subscriber B 20. The telephony servers 40,

516 306 are ΑΧΕ-gears with modifications as described below with reference to FIG.

2nd They perform the equivalent role of a local exchange or terminal in a standard wired telephony system, with the significant difference that those who switch utilize the IP-based data packet network. Each telephony server 40, 50 can also handle wired switching calls and thus interact with a traditional wired network, as illustrated by the rendered interface of the telephony server 40 to a wired access network 80 and the interface of the telephony server 50 to a wired SS7 network 90.

Also, a telephone number server 60 is connected to the data network 30. The telephone number server 60 is essentially a database containing reference tables for converting telephone numbers identifying a subscriber to an IP address or addresses for the host telephone server 40, 50 for the subscriber to which the call is to be connected. . It has been implemented on a standard UNIX DNS server with BIND software utilizing standard DNS record types, which converts e.164 numbers into IP addresses for network telephony endpoints in the network. Accordingly, the input to the reference table is an e.164 destination (B number) or part of this number. The output signal may be an IP address or addresses indicating the host telephone server for the subscriber in question.

The telephone number server 60 contains only the addresses of the subscribers connected to its own network 30. The IP addresses are defined in advance and each subscriber or each physical access to the IP network is assigned an address. The IP address preferably identifies a UDP port, which may be a fixed relationship to a time slot it utilizes. For some UDP ports, the routing information contained in an IP data packet may include the UDP port and the IP address of the port.

Also, a number of telephony resource devices 70 are connected to the data network in various positions. These are general telephony resources utilized by a telephony server 40, 50 to support a call. Typical resources include, but are not limited to answering machines, conference call devices, tone transmitters / receivers, tone detectors, voice messaging systems, and echo cancellers. It is advantageous that these resources are centrally located in

516 306 rather than being coordinated with a telephony server. This would significantly increase their utilization levels and make them more cost effective. However, since the telephone resources of the above type are already available in a conventional one

AXE switching retains these resources in the telephony server 40 in the preferred embodiment of Fig. 1.

Fig. 2 shows a schematic representation of a telephone server 40. The elements of the telephony server 40 comprise conventional switching hardware 41 for securing wired connections in the connected wiring. A number of terminals 42 are connected to the switching hardware and normally include a T1 transmission service terminal and an E1 transmission service terminal. Each terminal is connected to an associated wired network, in Fig. 2 represented by the network 80. The switching hardware 41 is further connected to an intermediate network line 43, which includes a PCM / IP adapter for converting a PCM-encoded 64 kbps channel into IP data packets for transmission via the IP-based network and vice versa. The PCM / IP converter 43 essentially packs the 64 kbps bitstream intended to be transmitted over the data network 30 in IP-compatible data packets or datagrams and similarly packs data packets arriving at the telephony server 40. Further, an IP switching controller 44 is provided to secure the virtual switching formed by the data network 30 for calls utilizing the data network. The IP switching controller 44 has essentially a logical view of the IP network 30 as an electron coupler with the input and output points of the coupler constituting the various media intermediate network lines 40 of the telephony server 40, such as the interconnect line 43, and the subscriber access nodes 11, 12. The IP switching controller 44 needs access to hardware for it to communicate with and control the various media intermediate network lines, but it does not need to include any hardware per se but rather the same IP network hardware interface as the intermediate network line 43. The telephony server further includes a controller 45 for controlling the the coupling function of various elements 41, 43, 44. The controller 45 also handles call setup and sends the required messages according to the agreed user / network protocol, such as V5.x or DSS1, to the subscribers 10 handled by the server 40. The controller 45 includes a routing function for handling all types of call transmission paths, i.e.

516 306 within the wired network, calls transmitted between the wired network and the data network, and calls transmitted within the data network.

As mentioned earlier, a number of telephony resources are available in the conventional AX exchange. These are represented in the telephony server 40 by the block 46. The occupancy of these resources for a call is controlled by the controller 45. The resources are then diverted to the call via switching hardware 41 or wired transmission paths or the interconnection line 43 if the call includes a subscriber on the data packet switched network 30.

The various elements of the telephony server 40, represented by the hardware 41 and the terminals 42, are needed only to enable a voice path to be routed through the telephony server to a wired network. If the call is connected between two subscribers 10,20 connected to the IP network, these elements will not be part of the communication. The interconnect line 43 is also used for interconnection calls, but it may also be required to generate a voice path when the telephony resources 46 are located as illustrated in the telephony server 40. Thus, those skilled in the art will recognize that these elements need not be part of a telephony server that does not serve as an interface. to a wired network and that does not include any telephony resources.

The network further includes a TNS resolver 46 to handle the interface between the telephone server 40 and the telephone number server 60. It establishes a connection with the telephone number server 60 when the telephone number server 60 is to be activated. Preferably, it will also handle the administration of the IP addresses to the telephone number server 60 such as defining new numbers and to update the telephone number server 60 when subscribers are switched on or disconnected. Although the management functions of the TNS resolver 46 are included in the telephony server 40, those skilled in the art will recognize that a separate network management node remote from the telephony server 40 could perform this function.

516 306

The access node 11 is shown in Fig. 3. A number of subscribers A, C and D are connected to the access node 11. The subscribers access this node 11 in the traditional wired way, that is via a two-wire PSTN terminal for the subscriber A, an ISDN BA (basic rate access, 2B + D) terminal for subscriber C or an ISDN PRA (primary rate access, 30B + D) terminal for subscriber D, which has been illustrated as a PABX. The PSTN and ISDN BA terminals are connected to a signaling function 12, which essentially performs the task of a conventional V5.x access node. In this function 12, the signaling is collected to or from the two-wire subscriber and transmitted into or out of a 64 kbps signaling channel with 2 Mbps pulse code modulation (PCM). A PCM / IP converter 13 is coupled to this signaling function 12 and packs the 64 kbps channel into the IP data packet, directing these data packets to the correct destination in the IP network. This routing is performed under the control of the telephony server 40. The PCM / IP converter also performs the reverse conversion and unpacks the IP data packets to a 64 kbps bit stream. The primary frequency access interface (PRA) is also provided with a PCM / IP converter 14 which performs a similar function. Although these converters 13, 14 are depicted as separate units in FIG. 3 the person skilled in the art will recognize that the same functional blocks can perform packing, unpacking, and routing functions for both interfaces.

The various forms of signaling that occur between the various elements over the data network 30 are summarized in the diagram of Fig. 4. All protocols are carried by IP. The superimposed protocols are preferably 'standard' or commonly used protocols that allow network elements to be exchanged for equipment from different makes. The so-called standard protocols are preferably well-established ITU protocols, which are easily understood by the network-connected network elements of the opposite side. This also allows the maintenance of the various operations supported by commonly used protocols. Accordingly, the signaling between the access node 11 and the telephony server 40 in call setup and for communicating routing information to the access node 11 utilizes a traffic protocol A] such as V5.x or the like over UDP. For ISDN primary frequency access, DSS1 is utilized over UDP. A connection protocol B] is also provided and can normally be V5.x BCC over TCP. Communication between two telephony servers 516 306 connected to the IP-based network utilizes a traffic protocol D], which is normally an SS7 ISUP over TCP. A connection protocol E] is also provided, which may be part of the ISUP or lie outside the ISUP. The telephony servers 40, 50 communicate with the telephone number server 60 to request the destination IP address associated with a particular b number using an appropriate protocol C], which is preferably a DNS protocol over TCP. Finally, two additional protocols are provided for communication between a telephony server 40, 50 and the telephony resources 70. These include a protocol F] for occupancy and control resources, and the F'J connection portion of F]. F] and F '] do not have to be standard protocols since the telephony resource nodes 70 are completely new elements that have no equivalent in a wired system. For the current data packet transport IETF protocol was used. The same goes for lower address resolutions and new network elements.

Before allowing calls over the access node 11, a signaling channel must be established between the signaling function portion 12 of this access node 11 or PABX and the host telephony server 40. This means establishing a LAPV5 and a LAPD data layer, respectively. An administrative procedure initiates an order from the telephony server 40 to the PCM / IP converter 13, 14 in question to start transmitting or receiving IP data packets containing information from the 64 kbps signaling channel in the 30B + D or V5.x interface. The PCM / IP converter 13.14 treats this signaling channel as a pure 64 kbps bit stream, but it can recognize the LAPV5 or LAPD frame format and transmit these frames as UDP / IP data packets. Obviously, the same handling of the signaling channel must be performed on both sides of the link. Thus, the PCM / IP converter in the intermediate line 41 of the telephony server 40 operates in the same way as the converter 13, 14. The access node 11 performs the normal startup procedures between a V5.1 access node or PABX and the host telephone server. The signaling to and from the telephone server 40 can then be transparent to the PCM / IP converters 13,14.

When a call is to be connected or disconnected, the PCM / IP converter 13, 14 receives additional orders from the host telephony server 40 for establishing or releasing

516 306 roads. Establishing a path involves directing the agreed 64 kbps bitstream between the telephony server 40 and PABX or the signaling function 12, packing it into the UDP / IP data packet or extracting it from the UDP / IP data packet, and directing it or it to the destination specified of telephony server 40.

If telephony resources 70 are to be utilized in a call initiated by subscriber A to subscriber B, then these resources must first be occupied by the host telephony server 40 using protocol F] as described above. Then, the resource function is routed through the access network 11 which connects the calling subscriber A. Thus, if a tone transmitter has been classified as a resource, 'tone' data packets should be sent from the telephony resource node 70 to subscriber A, while voice data packets are exchanged between subscribers A and B during a conversation.

In the case illustrated in Fig. 1, where telephone resources are available as part of the telephony server 40, a 'voice path' must initially be established between the access node 11 of the subscriber and the telephony server 40 when the call is set up at digital reception to enable the telephony resources to be made available to call. This is indicated by the dotted line between the telephony server 40 and the access node 11 in Fig. 1. A similar path shall be connected for voice data packets if subscriber A is to make or receive a call to or from a subscriber connected to the wired access network 80.1. Figure 1 indicates the UDP / IP voice data packets exchanged between subscriber A and subscriber B in a similar manner. with a dashed line over the data packet switched network 30.

As indicated above with reference to Fig. 1, the telephone number server 60 contains only address information for endpoints within the network to which it is connected. If a call is destined to a subscriber connected to neighboring networks, the telephone number server 60 in the network 30 will have no destination address. Fig. 5 illustrates the situation when calls cross two separate IP-based data packet switched networks, the network A and the network B. In fig. 5 the elements of network B are considered to be similar to those of network A and have been designated by the same reference numerals modified by a prime character. IN

516 306 FIG. 5, IP network interface network lines 61.61 'are connected to each data network 30, 30'. These intermediate network lines 61.61 'may be separate elements in the network or they may form part of the associated telephony server 40.40'. The network interface network lines 61, 61 'handle communication between different IP-based networks or IP between IP-based networks operated by different operators. The way this happens depends on how the operators of neighboring networks agree to cooperate.

According to a first embodiment of the present invention, each telephone number server 60,60 'in a first network 30 holds the addresses of network interface network lines 61, 61' to neighboring networks. When a call from network A is destined for network B, the telephone number server 60 in network A will provide the telephony server 40 in network A with the IP addresses of the network interface network line 61 'to the requested end or transit network, in this case the network B. When the call arrives at the end network via the connection point between networks A and B, the end network interface network line 61 'will request that its telephone number server 60' determine where the call is to be terminated. Thus, a call between the networks elicits two calls to the telephone number servers 60, 60 ', namely a request from the telephone server 40 to the telephone number server 60 in the network A and a request from the network interface network 61' to the telephone number server 60 'in the network B.

In a second embodiment of the present invention, the telephone number server 40 of a network does not contain the addresses of the network interface network lines 61 'in foreign networks but instead keeps the addresses of the network interface network lines 61 in its own network. The telephone number server 60 thus contains only the addresses of real endpoints in its own network. The network interface network lines 61 then store the addresses of neighboring network interface network lines entered by the operators of these networks. All calls between neighboring networks would then be forced through these network interface network lines.

Figs. 6 - 8 show examples of network interface network lines 61.61 'operating in accordance with the latter embodiment. In these embodiments, the network interface network lines 61.61 'are separated from the telephony server 40.40' and are intended for both incoming and outgoing calls. Similar parts are indicated by the same reference numerals in these three figures.

Fig. 6 shows two network interface network lines 61.61 'NGB1 and NGB2, which are interconnected with a wired interface. A first network interface network line 61 belongs to network A owned by a first operator. The second network interface network line 61 'belongs to a network owned by a second operator. The boundary between the two IP-based networks A and B is illustrated by the dashed line. Each network interface network line 61, 61 'includes a signaling intermediate network line SG1 611, SG2 611' for exchange of standard telephony protocol messages and a switch signaling protocol CONSIG, and a message intermediate network line MG1 612, MG2 612 'for the exchange of voice data. The transition from an IP-based format to a wired format increases the delay of the speech stream, but the interface is relatively easy to monitor for statistical functions and control or the like. The interface between the network interface network lines 61, 61 'is a standard telephony interface such as ISUP over E1. The signaling intermediate network line 1 SG1 receives an address for speech from the local telephony server 40 consisting of an IP address and UDP port in the CONSIG protocol. This address points to the A subscriber page and is used by MG1 to route the IP data packets originating in the subscriber B and received via a time slot on the wired interface to the subscriber A. The address of the voice stream, which originated in subscriber A, to subscriber B is generated by the network interface network line NGB2 61 'after it has polled its local telephone number server. It is transmitted to NGB1 61 by the signaling intermediate network line SG2 via the CONSIG protocol. The voice stream from A to B was transmitted from A in data packets with a destination address for MG1. MG1 unpacks the speech stream and introduces it in a time slot. MG2 repackages the voice stream in IP data packets with the destination address of B. The reverse process occurs for the speech stream from B to A. In this embodiment, the IP / UDP addresses of the two networks do not need to be coordinated. The network interface network lines NGB1 and NGB2 need only be associated via CIC time slots per call basis.

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Thus, in the embodiment shown in Fig. 7, the speech stream passes through the IP network as illustrated by the network of connected conductors R. The network interface network lines 61,61 'thus do not comprise any message intermediate network line but only have one signaling intermediate network line 61 611. Like the previous embodiment, the 6 ISUP / TCP messages and CONSIG / TCP messages (both over IP) with elements within their own networks. However, these messages are also exchanged in the interface between the network interface network lines 61.61 ', which is an IP-based interface. The CONSIG protocol is not completed. The CONSIG protocol transports the addresses of A to B and B to A. The voice data packets are transmitted directly through the IP-based network without control of the network interface network lines 61.61 '. However, the interface between the two adjacent networks A and B is designed so that all voice traffic between operating networks always goes between an assigned pair or assigned group of conductors. This is illustrated by the solid lines between the conductors R representing possible paths. In this way, measurement and monitoring functions can still be performed in these assigned conductors.

In the third embodiment shown in Fig. 8, the arrangement of the network interface network lines 61.61 'is very similar to that shown in Fig. 7, except that the operators of the networks A and B cannot impose any physical constraints as to the way in which the voice traffic will arrive. to take within the network. The IP-based network can also be owned by a separate operator. In order for the operators of the network where the call originates and terminates, to be able to limit the voice traffic to pass a certain point in the IP network, for example for enabling measurement, a routing control function has been introduced in the network interface network lines 61,61 '. The conductor may lie outside the network interface network lines 61.61 'as shown in FIG. 8 or be integrated with the network interface network lines 61, 61' in the manner applicable to the message interface network line 612 of FIG. 6. The signaling intermediate network line 611 will then not only bypass the endpoint addresses in the CONSIG protocol. Instead, an address received by the signaling intermediate network line 611 will be transmitted to a new address transmitted (as a representative) and a control interface illustrated by a thick line in Fig. 8, and protocol G will command a conductor to feed speech stream, which arrives on one interface, to be routed through another interface.

The arrangement of the telephone number server 60 essentially produces a plan, not hierarchical structure, where neither the IP addresses nor e. 164 numbering plan need to be coordinated with other network administrations. The only coordination needed is for the inter-coordination points or network interface network lines 6 to other networks. This structure thus exploits the advantages of IP connection and with the more well-structured e.164 numbering plan, where each network administrator has more freedom in defining the numbering.

The arrangements described above provide a means for bringing an IP-based network closer to the subscriber by replacing the underlying wired transport network with an IP-based data packet-switched network while maintaining costly elements in a circuit-switched network.

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