WO2004102941A1 - Systeme et procede d'acheminement de communications telephoniques d'un telephone classique par le biais d'un reseau de donnees - Google Patents

Systeme et procede d'acheminement de communications telephoniques d'un telephone classique par le biais d'un reseau de donnees Download PDF

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Publication number
WO2004102941A1
WO2004102941A1 PCT/CA2004/000396 CA2004000396W WO2004102941A1 WO 2004102941 A1 WO2004102941 A1 WO 2004102941A1 CA 2004000396 W CA2004000396 W CA 2004000396W WO 2004102941 A1 WO2004102941 A1 WO 2004102941A1
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WO
WIPO (PCT)
Prior art keywords
subscriber
call
interface unit
data network
data
Prior art date
Application number
PCT/CA2004/000396
Other languages
English (en)
Inventor
Anatoly Abelev
Barry Cecil King
Harry Allen Trefry
Mario Ivanic
Michael Alievsky
Richard George Todd
Stephen William Tunks
Original Assignee
Easylink Networks Inc.
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Easylink Networks Inc. filed Critical Easylink Networks Inc.
Publication of WO2004102941A1 publication Critical patent/WO2004102941A1/fr

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M7/00—Arrangements for interconnection between switching centres
    • H04M7/0024—Services and arrangements where telephone services are combined with data services
    • H04M7/0057—Services where the data services network provides a telephone service in addition or as an alternative, e.g. for backup purposes, to the telephone service provided by the telephone services network
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04Q—SELECTING
    • H04Q3/00—Selecting arrangements
    • H04Q3/64—Distributing or queueing
    • H04Q3/66—Traffic distributors
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04Q—SELECTING
    • H04Q2213/00—Indexing scheme relating to selecting arrangements in general and for multiplex systems
    • H04Q2213/13034—A/D conversion, code compression/expansion
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04Q—SELECTING
    • H04Q2213/00—Indexing scheme relating to selecting arrangements in general and for multiplex systems
    • H04Q2213/13138—Least cost routing, LCR
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04Q—SELECTING
    • H04Q2213/00—Indexing scheme relating to selecting arrangements in general and for multiplex systems
    • H04Q2213/13389—LAN, internet

Definitions

  • the present invention relates to telecommunications systems and in particular to the routing of telephone communications.
  • the invention is especially applicable to a system and a process for routing telephony commumcations from a conventional ("PSTN") telephone set through a public data network, such as the Internet, to terminate, optionally, at another conventional "PSTN" telephone; and to a subscriber interface unit for use therewith.
  • PSTN public data network
  • the invention is applicable whether access to the network is narrowband or broadband.
  • the first class of conventional routing system usually used by large companies with private communications links, uses "re-diallers" .
  • Re-diallers re-route specific calls through a separate transmission path, e.g. a data network, using a real-time transport mechanism.
  • This system requires the re-dialler to have access to, and knowledge of, a private wide area data network, which is a significant disadvantage.
  • the second class of conventional routing system provides real-time voice communications from and to computer systems which have analog-to-digital conversion hardware and appropriate software; the user using a headset connected to his computer. Recent modifications have allowed the use of a standard telephone set attached to the
  • IP Internet Protocol
  • United States patent number 6,205,135 discloses a routing system using an alternate access platform located at a local exchange or, perhaps, at another location within the PSTN network that is 'further away from the subscriber. Limitations of such a system include the fact that the platform is expensive and hence not cost effective unless a large number of subscribers access the platform to make Internet-based telephone calls. Moreover, if the company providing the Internet-based telephony services does not also own the local exchange, it will be charged a so-called collocation fee, leading to increased cost to the subscriber,
  • European patent application No, 1 061 728 Al discloses a least-cost routing system which is similar to those of the first class described above, but which uses a public data network, specifically the Internet, rather than a private network.
  • Each subscriber has an IP router which establishes connections between the subscriber's telephone and either a telephone network provider or the Internet. When installing the IP router, the subscriber compiles his individual telephone directory with the telephone numbers and corresponding IP addresses of all parties he may wish to call.
  • the router uses it to access the individual telephone directory and determine whether there is a corresponding IP address, If there is none, the router uses least-cost-routing based upon the cost information to route the call over the conventional telephone network, If there is a corresponding IP address, however, the router determines whether the estimated cost of the telephone connection via an Internet connection is more favourable than that of a normal telephone call and, if it is, routes the call as an IP call.
  • An object of the present invention is at least to ameliorate the disadvantages of such known systems, or at least provide an alternative.
  • a method of routing telephone calls from a subscriber telephone in a telecommunications system comprising the Public Switched Telephone Network and a public data networ , for example the Internet, to a called party, the system including a local central office, a data network entrance node, a data network exit point, a subscriber interface unit located at the subscriber premises and connected between the subscriber telephone and the local central office, a called party interface unit connected to the data network exit point, a call server provisioned with data and software for routing of data calls between the data network entrance node and the datanetwork exit point, the call server dataincluding destination numbers and corresponding network addresses of called party interface units that are capable of converting signals between PSTN and data formats, and a management server having software and data including rules for identifying classes of calls of a particular subscriber which should be routed via the data network, the subscriber interface unit having means for accessing said rules, the method comprising the steps of: at the subscriber interface unit,
  • step (iii) if it is determined in step (ii) that the call not within the Specified class, routing the call to a local central office for processing as a conventional PSTN telephone call; (iv) if it is determined in step (ii) that the call is within a specified class and should be routed through the data network, connecting to the call server (38) and supplying to the call server subscriber interface unit identification and the called party destination number; at the call server,
  • Preferred embodiments of the present invention provide individual subscribers with a subscriber interface unit capable of deciding whether to route a particular telephone call via the PSTN or a public data network on a per call basis using network selection ailes, conveniently supplied by the subscriber's service provider, that identify different classes of calls that should be routed via the data network, The decision process compares some of the dialled digits with the network selection rules to determine the class of call.
  • the interface unit may compare only enough of the most significant digits of the called party number to enable it to determine whether or not desired call falls within one of the classes defined by the rules, Hence, it is not necessary to use the entire destination number.
  • neither the subscriber nor the subscriber interface unit needs to have knowledge of the data network address of the called party. Moreover, the subscriber does not need to pre-program the telephone for data network calls by entering a separate list of destination numbers to which calls may be made via the data network.
  • a telecommunications system comprising the Public Switched Telephone Network and a public data network, for example the Internet, and means for routing calls between a subscriber telephone set and a called party, the system including a local central office, a data network entrance node, a data network exit point, a subscriber interface unit located at the subscriber premises and connected between the subscriber telephone and the local central office, a called party interface unit connected to the data network exit point, a call server provisioned with data and software for routing of data calls between the data network entrance node and the data network exit point, the call server data including destination .numbers and corresponding network addresses of called party interface units that are capable of converting signals between PSTN and data formats, and a management server having software and data including rules for identifying classes of calls of a particular subscriber which should be routed via the data network, the subscriber interface unit having means for accessing said rules, wherein: the subscriber interface unit comprises, means for (i) detecting digits dialled by the
  • the connection to the network node is initiated while the dialled digits are being collected and the routing decision made, so as to reduce waiting time for the subscriber.
  • Broadband embodiments of the invention may provide so-called "call waiting:", i.e., maintaining PSTN and VOIP calls simultaneously and allowing the subscriber to select one or the other without terminatmg.e.ther, According to a third aspect of the present invention, there is provided a subscriber
  • the subscriber interface unit having a first port for connection to a subscriber telephone set, a- second port for connection to a subscriber line, means for detecting dialled digits received via the first port and using at least
  • the subscriber interface unit may perform analog-to-digital and digital-to-analog conversion of signals before and after the conversion to and from the • 15 data signals, Likewise, if the called party equipment is analog, the second interface unit may provide D-to- A and A-to-D conversion in addition to conversion to and from the data signal format,
  • the subscriber interface unit and the second interface unit may comprise means for converting from subscriber or called party 20 signal format to packetized data signals, and vice versa.
  • the subscriber interface unit may include a conventional modem for converting the data signals into a format suitable for transmission to the local central office as a PSTN signal
  • a conventional modem for converting the data signals into a format suitable for transmission to the local central office as a PSTN signal
  • the conventional modem could be omitted and means provided for connecting the data
  • 25 conversion means e.g. codec, to a high speed modem.
  • the subscriber interface unit is installed at the subscriber' s premises between the telephone set and the second interface unit is a PSTN gateway device connected to the called. party equipment via the PSTN.
  • the second interface unit could be similar to the
  • the stored data accessed by the subscriber interface unit may be stored at the subscriber interface unit itself and periodically updated via the data networkby a management application server unit at a remote location.
  • the subscriber interface unit has a mechanical switch connected between an input port and an output port for connection to the subscriber telephone set and the subscriber line, respectively, the switch being biassed to connect the input port and output directly in the event of power failure to the subscriber interface unit.
  • the subscriber interface unit may have a port for connection to a high speed modem, in which case the conventional internal modem can be omitted.
  • the connection to the first network node is not taken down immediately in case the subscriber decides to make another call immediately. Rather, when a first call ends, the interface unit will monitor for the subscriber dialling new digits and, when such digits are detected, determine the routing for the new call, setting up a new communication session, if appropriate, with the call server to set up a new connection to the new called party.
  • a process for routing telephone calls including providing dial tone to a standard telephony device, collecting therefrom dialled digits representing a telephone number of a called party and using at least some of the dialled digits to access stored data and determine whether or not to route the call over a public data network; connecting through the PSTN to a data network
  • ISP ⁇ connection point
  • the process is capable of routing telephony communications from a public switched telephone network (PSTN) over a public data network (Internet) and back to a public switched telephone network (P STN) .
  • PSTN public switched telephone network
  • Internet public data network
  • P STN public switched telephone network
  • Figure 1 is a schematic diagram of a first embodiment of the present invention for routing telephone calls within a public data network
  • FIG 2 is a detail schematic view of a narrowband subscriber interface unit shown in Figure 1;
  • Figure 3 is a flowchart, illustrating a first stage in the setting up of a call using the subscriber interface unit;
  • Figure 4 is a flowchart illustrating a second stage in the setting up of the call;
  • Figures 5 A and 5B are flowchart sections illustrating the ending of the call in different ways
  • Figure 6 illustrates a second embodiment of the invention for routing telephone calls through a public data network using a broadband connection
  • Figure 7 illustrates a broadband subscriber interface unit of the system shown in
  • Figure 8 illustrates a first part of a sequence of setting up of a call using the broadband subscriber interface unit
  • Figure 9 illustrates a second part of the sequence
  • Figure 10 is a flowchart illustrating the ending of a call in different ways using the broadband subscriber interface unit
  • FIG. 11 illustrates the handling of an inbound call by the broadband subscriber interface unit
  • Figure 12 is a flowchart illustrating in more detail routing decision steps of Figures 3 and 8.
  • FIG. 1 illustrates schematically deployment of a narrowband or "dial-up" embodiment of the invention in the public telecommunications system comprising a Public Switched Telephone Network (PSTN) and a public data network, e.g., the Internet.
  • PSTN Public Switched Telephone Network
  • the router/dialler unit 1 is installed at the subscriber's premises and is provided (leased or purchased) to enable the subscriber to use the service,
  • the station apparatus 18 of a third party (the Called Party) is shown connected via a subscriber loop 20 to a remote or destination central office 22 in the PSTN 16.
  • the local or originating central office 14 and the remote or destination central office 16 are interconnected via the PSTN, as represented by broken line 24.
  • the router/dialler unit 12 stores the telephone number of a
  • the HOME ISP 26 accessible via the PSTN and data link 28,
  • the HOME ISP 26 which is provisioned for Voice-over-Internet Protocol (VoIP)
  • VoIP Voice-over-Internet Protocol
  • the gateway 32 is coupled via link 36 to the PSTN and therefore via the PSTN to the destination central office 22.
  • a call server 38 shown connected to the ISP 26 could be local to the ISP or at a remote location and accessed over the Internet or another data network.
  • the call server 38 is provisioned with a dialled-digits store and routing data and software enabling it to route calls to network nodes, including ISPs, in a known manner.
  • router/dialler unit 12 The data stored by router/dialler unit 12 is downloaded to router/dialler unit 12 during an initialization process carried out once it has been installed and connected, and then updated periodically, by a management application 40, namely software conveniently running on a personal computer acting as a server that is located on the premises of the ISP 26, or at a remote location on the network; and able to communicate with the ISP 26 using the usual protocols.
  • the router/dialler unit 1 is programmed with the telephone number of the
  • the management application unit 40 will download to the router/dialler unit 12 whatever information needs to be stored in the router/dialler 12 to enable it to determine whether or not a particular call should be a VoIP call or regular PSTN call.
  • the management application 40 is provisioned with feature software enabling it to ' ⁇ provide for Administration, Account Verification, Dialing Rule assignment/update and ' Router/dialler Setup. • The typical data that are downloaded by.
  • the management application unit 40 will include Network selection rules specifying that some calls are candidates for routing through the data network (VoIP calls) while others should be routed through the PSTN only. For example, international calls or other toll call s might be candidates for VoIP, whereas toll-free calls can be routed as PSTN-only calls. The actual selection would be at the discretion of the service provider, who might, for example, prefer to include toll-free calls as candidates for VoIP as well. Likewise, if local calls are free, as in most parts of North America, they too may be routed as PSTN-oniy calls.
  • the following table lists, as an example, a set of rules for making a routing decision between PSTN and VoIP networks,
  • the table is generated by management application unit 40 on a per unit basis and downloaded to ' the SPIU 12, during either the initialization or a configuration update, where it is stored in memory,
  • the SPIU 12 collects incomin digits from the subscriber's telephone and compares them against network selection rules in its memory, If the'destination number matches one of the rules, the call will be placed through the digital data network (VoIP); otherwise, if no match is found, the dialed number will be redialed to route the call via the PSTN,
  • VoIP digital data network
  • the following table shows an example of the network selection rules of calls that are to be routed via the VoIP (data) network, all other calls, by virtue of their "absence", being routed via the PSTN by default:
  • the router/dialler unit 12 When the subscriber begins to make a call, the router/dialler unit 12 will detect th dialled digits, consult the stored data (rules) and determine whether the call should be route as a regular POTS call via the PSTN or as a Voice-Over-Internet-Protocol (VoIP) call, an route the call accordingly. This happens without the subscriber being involved in the decision.
  • VoIP Voice-Over-Internet-Protocol
  • the router/dialler unit 12 determines that the call should be routed as a POTS call, it will simply route it via the central office 14 into the PSTN in the usual way (as represented by line 24).
  • the router/dialler unit 12 determines that the call should be routed as a Voice-over- Internet-Protocol (VoIP) call
  • the router/dialler converts it into a data signal with the usual header etc, and routes it to the Internet in the usual way via HOME ISP 26, having first set up the connections by exchanging SIP messages with the call server 38.
  • the call server 38 detects the called party destination number in a "SIP Invite" message frorr the router/dialler 12, uses it to determine the closest gateway network node for th. destination, then establishes a connection to PSTN gateway node 32. Once the connectio is established, the gateway node 32 supplies the gateway parameters, e.g.
  • the call is converted into a PSTN (analog voice) call again and routed via the PST to the destination central office 22, which completes the call in the usual way to the called party 18.
  • PSTN analog voice
  • the router/dialler 12 comprises afirst port 42 to which the subscriber's telephone set 10 is connected, a mechanical relay 44, a microprocessor 46, a DSP/CODEC 48, a Hayes-compatible modem 50 (specifically a V90 56 kb/s mode ), an PSTN port 52 which is connected to the subscriber line, a voltage regulator unit 54 connected to DC input socket 56 for connection to a separate power supply (not shown in
  • the voltage regulator 54 supplies the various components at the appropriate voltage.
  • the SLIC 58 provides the
  • the DSP/CODEC 48 provides digital-to-analog and analog-to-digital conversion, together with voice compression and decompression to ensure that the data signals are within the capability of the modem 50,
  • the processor 46 controls the relay 44 and other components, as necessary, and controls the setting up and taking down of PPP communications with the HOME ISP 26 and SIP communications with the call server 38, In addition, the processor 46 converts the digitized voice signals to data signals, specificall VoIP signals, and vice versa,
  • the subscriber' s telephone set 10 is connected via input port 42 and mechanical relay 44 either to code 48 or directly to output port 52, If there is a power failure, or fault, the relay 44, by default, connects the telephone set 10 to the output port 52 via line 61 so the subscriber can make conventional calls, especially emergency calls, without involving the modem, etc. Moreover, prior to initialization of the recently-installed router/dialler unit 1 , the processor 46 sets the relay 44 to connect the subscriber set directly to the output port 52.
  • the initialization procedure is triggered when the subscriber lifts the receiver of the phone device 10 and dials a predetermined initialization sequence. This action will initiate a VoIP connection setup with a call center representative. The call is then terminated at the call center and a customer service representative verifies all applicable customer profile information with the end user, Once that has been completed, the customer service representative initiates a procedure that provisions the management application server 5 40 with appropriate network selection rules and the call server 38 with authentication data (if required). The management application 40 then supplies the network selection rules, address of call server, configuration data, and so on, to the router/dialler 12 using a proprietary protocol. The received information is stored in the permanent unit memory and updated periodically by the management application 40, as necessary.
  • the processor 46 will set the mechanical relay 44 to couple the telephone set 10 to the DSP/CODEC 48, i.e., the default presumes that any call will be VoIP.
  • the processor 46 monitors the input port 42 so that, as soon as the subscriber picks up the
  • the processor 46 can detect the "off hook” condition.
  • the SLIC 58 detects the subscriber telephone set "going off-hook" in step 3.02 when the subscriber begins to make the call and reports the event to the processor 46 by way of the DSP/GODEC 48.
  • step 3 the processor 46 activates the modem 50 and, in step 3.04, detects via the modem 50 the usual dial tone supplied by the central office 14, In step 3.05, the processor 46 accesses its
  • processor 46 has not yet connected the subscriber to the central office 14 so the subscriber does not hear the usual central office dial tone, Instead, in step 3.06, the processor 46 causes DSP/CODEC 48 to generate a "placebo" dial tone and
  • step 3.07 the subscriber dials the telephone number of the called party 18 (step 3 ,08),
  • step 3.09 the processor 46 collects the dialled digits, instructing the DSP/CODEC 48 to discontinue the placebo dial tone on receipt of the first digit, and in step 3.10 stores the digits temporarily in a buffer. Meanwhile, in step 3.11, the processor 46 is completing the usual "handshake" process with the HOME ISP 26, supplying login identification and password as appropriate.
  • the processor 46 accesses its stored data, specifically its "call routing rules", to determine whether the dialled digits represent a destination number to which the call should be routed as a PSTN-oniy call or as a VoIP call. If the rules specify that the call should be a PSTN-oniy call, in step 3.13 the processor 46 aborts the establishment of the connection to the HOME ISP 26, or terminates it if completed, then, in step 3,14, retrieves the dialled digits from its buffer and relays them via mode 50 to the central office 14.
  • the processor 46 accesses its stored data, specifically its "call routing rules", to determine whether the dialled digits represent a destination number to which the call should be routed as a PSTN-oniy call or as a VoIP call. If the rules specify that the call should be a PSTN-oniy call, in step 3.13 the processor 46 aborts the establishment of the connection to the HOME ISP 26, or terminates it if completed, then, in step 3,14, retrieves the dialled
  • step .15 the processor 46 switches relay 44 to connect the subscriber' s telephone set 10 to line 61 and port 52, thereby connecting the subscriber directly to the central office 14 to hear the usual central office ringing and other call progress tones (step 3.16).
  • the call then continues as a regular PSTN-oniy call as indicated by step 3.17. If, in step 3.18, either party terminates the call, step 3.1 returns the processor 46 to scanning for originations (step 3.01).
  • step 12.01 sets a digit counter to zero and step 12,02 retrieves the corresponding digit from the buffer (see step 3.10 of Figure 3).
  • step 12.03 the digit is compared with the appropriate digit from the network selection rules list. It should be noted that the numbers listed in the network selection rules are sorted in ascending numerical order. If step 12,05 determines that there is no match, step 12,06 causes the call to be placed via the PSTN,
  • step 12.08 determines whether or not the current digit is the.last digit (what constitutes "last" being determined by the network selection rules), If it is not, step 12.07 increments the digit counter and the processor repeats steps 12.02 to 12.08.
  • step 12.08 determines that the current digit is the last digit
  • step 4.01 determines that the appropriate dialling rule has been met, i.e., the destination number has been collected and validated against the rules, decision step 3.1 ( Figure 3) determines that the call should be routed as a VoIP call, and the router/dialler 12 performs the functions illustrated in Figure 4. As shown in step 4.01, the processor 46 causes DSP/CODEC 48 to .
  • step 4.03 the processor 46 accesses its stored data to determine the network name (address) of call server 38 and, using Session Initiation Protocol (SIP), initiates a call to it by sending it a "SIP Invite" message containing its unique identifier and the dialled digits.
  • SIP Session Initiation Protocol
  • step 4.04 accesses its call server authentication data to confirm that the router/dialler 12 is used legitimately.
  • the call server authentication data is updated periodically by the management application 40 (Figure 1) with up to date authentication data.
  • the call server 38 uses its routing data to identify the appropriate gateway based upon this routing information.
  • the routing information could be provided either by network administration (static routes) or automatic gateway location services (gatekeepers, ENUM and so on).
  • step 4.06 the call server 38 establishes SIP-based communications with the PSTN gateway 32.
  • step 4.07 the PSTN gateway 32 completes the call to the called party 18 and notifies the call server 38 which supplies a valid termination message to the processor 46, RTF then is used to set up a bearer connection between the router/dialler 12 and the gateway 32 and ringing tone from the destination central office 22 is conveyed to the processor 46.
  • step 4.08 the processor 46 conveys the ringing tone via the DSP/CODEC 48 and SLIC 58 to the subscriber telephone set 10 (step 4.09).
  • step 4.10 the called party answers, the call server 38 notifies the processor
  • step 4.11 causes the DSP/CODEC 48 to discontinue the ringing tone.
  • step 4.12 the DSP/CODEC 48 begins A-D conversion and compression of the voice signal from the subscriber set 10 and the processor 46 converts the compressed digital signal to VoIP data for transmission by the modem 50 to the gateway 32.
  • the VoIP data will be converted to PSTN format (step 4.13) and routed via the PSTN to the called party in the usual way.
  • the PSTN gateway 32 digitizes the called party voice signals and converts them to VoIP sends them via the ISPs 30 and 26 to the router/dialler unit 12 where the processor 46 converts them from VoIP to digital and the DSP/CODEC 48 converts the digital signals to analog signal and routes the analog signal to the subscriber telephone set 10 ' viathe relay 44.
  • the call proceeds in this manner as indicated in step 4.14, and the processor 46 and gateway 32 monitor for one party or the other ending the call by "going on-hook".
  • step 5.01 constitutes one of the parties ending the call by "going on-hook" (this is the PSTN call on-hook).
  • step 5.02 If it is the called party- 18 who - terminates the call and goes "on-hook” first, in step 5.02 the gateway 32 recognizes the "on- hook” condition and by a SIP message notifies the call server 32 which, on receipt ofthe notification in step 5.03 relays it to the processor 46 in router/dialler- 12. On receipt ofthe notification in step 5,04 the processor 46 causes DSP/CODEC 48, in step 5.05, to provide a placebo dial tone to the subscriber telephone set 10. The S ⁇ P-based VoIP call has now been taken down, ⁇ The processor 46 does not immediately terminate the connection to the HOME ISP
  • step 5.06 sets a timer and waits for a predetermined time while monitoring for (i) timer "time-out”; (ii) new dialled digits received, indicating that the subscriber has decided to make another call immediately, without hanging up the handset; and (iii) "off- hook” detected, to be explained later. If the timer times out, in step 5.07 the processor 46 will cause the DSP/CODEC 48 to discontinue the placebo dial tone and then, in step 5,08, will "tear down" the connection to the ISP, in step 5.09 wait for the subscriber to hang up and then proceed to step 3,01 ( Figure 3) to wait for originations. Discontinuing the placebo dial tone will force the subscriber to hang up and go off-hook again should the subscriber wish to make another call.
  • step 5.11 will take the path of "new digits dialled", in which case, in step 5.12, the processor 46 will cancel the timer, in step 5.13 cause the DSP/CODEC 48 to discontinue the placebo dial tone and then proceed to step 3 ,09 ( Figure 3) to collect digits, then continue as before to set up the new call. In this case, the processor 46 will continue to maintain the connection to the ISP 26.
  • step 5.11 the processor 46 proceeds to cancel the timer in step 5.14 and then go to steps 5.07 to 5.09, causing the DSP/CODEC48 to discontinue the placebo dial tone, tearing down the connection to the HOME ISP 26 and then proceeding to step 3.01 ( Figure 3) to scan for originations.
  • step 5.15 the processor 46 detects the "on-hook” condition and in step 5.16 notifies the call server 38 using SIP, On receipt o the notification, in step 5.17, the call server 39 will tear down the SIP connections to both the processor 46 and the gateway 32.
  • the processor 46 does not terminate the connection to the HOME ISP 26 immediately, however, in case the subscriber decides to make another call -immediately. Instead, the processor 46 goes to decision step 5.18, sets the timer and waits, while monitoring for the subscriber to go "off hook” again immediately and start dialling new digits or for the time-out to occur,
  • step 5.19 the processor 46 terminates the connection to the HOME ISP 26, then returns to step 3.01 of Figure 3, i.e., resumes scanning for originations.
  • decision step 5.18 takes the "off-hook detected” path, in which case, in step 5.20, the processor 46 will maintain the connection to the HOME ISP 26 and cancel the outstanding timer, then go to step 3.06 ( Figure 3) to cause DSP/CODEC 48 to supply the placebo dial tone while the processor 46 collects dialled digits in step 3.09 and so n,
  • the call server 38 will record when various events occur.
  • the processor 46 When the router/dialler 12 is in its "wait" condition, the processor 46 is monitoring not only the subscriber set 10 via the DSP/CODEC 48 and SLIC 50 for initiation of an outgoing call but also, via modem 50, the line 61 for indications of an incoming call. If the processor 46 detects an incoming call, it merely switches the relay 44 to connect line 61 directly to the subscriber's set 10. The processor 46 monitors line 61 to detect completion ofthe call at which point it causes relay 44 to return to its original state,
  • the latter can communicate directly with the call server 38, i.e., without going through gateway 32, and itself convert from VOIP to analog.
  • the codec units 48 in the two router/diallers 12A and 12 ' will handle the A-to-D conversion and compression, and vice versa, as appropriate. It would also be possible to complete the call to a called party with a SIP telephone or other compatible device elsewhere on the Internet, as indicated by optional ON NET EXTERNAL box 67.
  • FIG. 6 A second embodiment ofthe invention, which used a broadband router/dialler, will now be described with reference to Figures 6 to 12.
  • Components ofthe broadband system shown in Figure 6 which are the same as component in the embodiment of Figures 1 and 2 have the same reference numerals.
  • the subscriber telephone set 10 is connected to a broadband router/dialler 12A which differs slightly from the narrowband router/dialler 12 of Figures 1 and 2.
  • the router/dialler 12A has three • ports, a first port 42 for connection to the subscriber telephone set 10, a second port 52 for connection to the regular telephone line for transmission ofPSTN-only analog telephone calls, and a third port 62 for connection via a high speed modem 63 (see Figure 6) to a broadband subscriber line.
  • the router/dialler 12A also differs from router/dialler 12 in that it has no Hayes- compatible modem; instead it has a Data Access Arrangement (DAA) unit 64 connected between the processor 46 and the output port 52 and a DTMF transceiver 65 connected between the processor 46 and the DAA 63.
  • DAA Data Access Arrangement
  • the DTMF transceiver 65 detects ringing tones from the central office and provides a means whereby the processor 46 can dial digits for transmission to the central office 14.
  • port 52 is shown connected via subscriber loop 66 to the central office 14 f the PSTN 16 and port 62 is shown connected via high speed modem 63 to HOME ISP 26.
  • the broadband connection is an Asynchronous Digital Subscriber Line (ADSL)
  • both the high speed modem 63 and the regular PSTN port 52 would be connected via the same subscriber loop 66 to the central office 14 where a digital subscriber loop access multiplexer (DSLAM) would separate the PSTN-oniy calls and the data calls and route them to the PSTN 16 and the HOME ISP 26, respectively.
  • DSL digital subscriber loop access multiplexer
  • the high speed or data connection could, of course, be by telephone line, cable or a wireless/satellite connection. In all cases, however, it is presumed that the router/dialler 12A does not need to set up a call to the HOME ISP 26 because it is in continuous communicatio with it.
  • step 8.01 the processor 46 scans the subscriber telephone port 5 42 and line 61 for originations. If, in step 8.02, the subscriber picks up the handset to make a call, in step 8.03 the processor 46 causes the DSP/CODEC 48 to supply a placebo dial tone to the subscriber set 10. On hearing the placebo dial tone in step S.04, the subscriber begins to dial the destination telephone number ofthe called party 18 (step 8.05). In step 8.06, the processor 46 collects the dialled digits and, in step 8.07, buffers them while determining the routing for the call in step 8.08, using the rules previously stored in the router/dialler 12A.
  • the management application 40 downloads to the router/dialler 12A network selection rules and other data and updates it periodically.
  • the detailed decision process carried out in step 8.08 is that illustrated in Figure
  • step S.09 the processor 46 opens a connection to the. central office via DTMF transceiver 65 and DAA 64 and transmits to the central office 14 the dialled digits from its buffer.
  • step 8,11 the processor 46 switches relay 44 to connect the subscriber set 10 directly to the line 61 and thence to central office 14, whereupon, in step 8.12, the subscriber hears the ringing tone,
  • step 8.13 the call proceeds as a regular PSTN ⁇ call until, in step 8.14, either party terminates whereupon, in step 8,15, the processor 46 returns to scanning for originations (step 8.01).
  • step 8.08 the processor 46 determines that the call should be routed as a data (VOIP) call
  • the processor 46 proceeds to step 9.01 and causes the DSP/CODEC 48 to supply to the subscriber set 10 an "On Internet" tone to be heard by the subscriber (step 9.02).
  • step 9.03 processor 46 accesses its stored data to determine the network name/address ofthe call server 38 and uses SIP protocol to initiate a call, to the call server 38, supplying its own unique identifier and the destination number.
  • step 9.04 the call server 38 accesses it authentication data to confirm that the router/dialler 12A is used legitimately.
  • the call server 38 uses its routing data to determine the network name/address of the appropriate PSTN gateway 32 and in step 9.06 establish a SD?-based connection to the gateway 32 which completes the connection via the PSTN to the called party 18. .
  • step 9.07 the destination central office 22 supplies the usual valid termination message which the gateway 32 relays, using SIP, to the call server 38 and the processor 46.
  • the processor 46 On receipt of the valid termination message, in step 9.08 the processor 46 causes the DSP/CODEC 48 to supply to the subscriber 10 a ringing tone.
  • step 9.09 the call server 38 detects that the called party 18 has answered and relays the information to the processor 46 which, in step 9.10, instructs the DSP/CODEC 48 to discontinue the ringing tone and allows the call to proceed,
  • step 9.11 the DSP/CODEC 48 digitizes and compresses the analog signal from subscriber set 10 and the processor 46 converts the resulting digital signal to VOIP format and relays the VOIP signals via DAA 64, high speed modem 63, HOME ISP 26,
  • the PSTN gateway 32 and gateway 32 to the called party 18.
  • the- gateway 32 converts the VOIP signal to analog form again before routing it to the called party IS.
  • the gateway 32 digitizes the analog signal from called party
  • the calls from broadband router/dialler 12 also could be made to a subscriber 18B or to a SIP phone ISC or other ON NET EXTERNAL device.
  • Figure 10 illustrates the sequence of operations when either ofthe parties ends the call, i.e, "goes on-hook" in step 10.01. If it is the called party 18 who terminates the call and goes "on-hook” first, the gateway 32 detects the event and issues a SD? message to notify the call server 38. In step 10,02, the call server 38 recognizes the "on-hook” condition and, in a similar SIP message, notifies the processor 46 in router/dialler 12 A. On receipt of the notification in step 10.03, the processor 46 proceeds to step 8.03 ( Figure 8 ⁇ to allow for further calls by the subscriber. At this point, the SIP-based connection is down.
  • step 10.04 the processor 46 detects the "on-hook" condition and in step 10,05 notifies the call server 38 using SIP.
  • step 10.06 the call server 38 receives the message to take down the SIP connection and proceeds to do so according to SIP procedures. The processor 46 then goes to step 8.01 ( Figure 8) to scan for originations.
  • the call server 38 will record when various events occur. Operation o the broadband router/dialler 12 A to handle an incoming call is relatively straightforward. As shown in Figure 11 , when the router/dialler 12A is in its "scan" condition, S tep 11.01, the processor 46 is monitoring the subscriber set 10 for initiation of an outgoing call while the processor 46, via DAA 64 and DTMF transceiver 65,. monitors the line 61 for indications of an. incoming call. When the processor 46 and DAA 64 present on "on-hook" condition, the central office transmits a "ringback" signal to the calling party and then the processor 46 in step 11.03 determines whether the subscriber set 10 is "on-hook". If it is, in step 11.04 the processor 46 connects the call to subscriber set 10 and in step 11.05 the ringing tone is heard by the calling party.
  • step 11.06 the processor 46 determines whether or not there is a VOIP call in progress. If there is not, the processor 46 loops back o repeat steps 11.05 and 11.06 to periodically retests both conditions until it is positive or the inbound call is abandoned,
  • step 11.06 determines that there is a VOIP call in progress
  • step 11.07 supplies alert tones for a predetermined period to the calling party. These alert tones, when heard by the subscriber, inform the subscriber that there is an inbound call waiting. To accept the inbound call, the subscriber must terminate the VOIP call and then accept the inbound call by "going off-hook".
  • the broadband subscriber interface unit 12A shown in Figure 7 could be modified, specifically by programming of processor 46, to provide various special telephony features, such as "call waiting" between both PSTN and VOIP calls. Because the processor 46 is connected to both the DAA 64 and the broadband connector 62, it is capable of maintaining both PSTN and VOIP calls and allowing the subscriber to switch from one to the other.
  • the rules are stored in the subscriber interface unit 12/12A and updated periodically by the management server 40, it is envisaged that thebroadband version o he subscriber interface unit could access the management server whenever it needed to access the rules, instead of storing them itself, and updating them periodically.
  • callers use a Voice-over- IP ' systerri to place outgoing calls over an ISP's network
  • Customers use their own telephone instrument, plugged into a router/dialler, to access the ISP's network.
  • the router/dialler connects to the ISP's network using a PPP connection when the customer places a call.
  • the ISP operates a number of PSTN Gateways which bridge voice calls between the ISP's network and the PSTN. Calls are processed by a Call Server, which performs authentication, gateway selection and, optionally, call logging.
  • a management application exists to enable the service provider's support personnel to configure new router/diallers or update gatewa . lists.
  • the public data network is the Internet
  • the invention could work with other data networks.
  • the above-described subscriber interface unit is separate from the subscriber's telephone set, it could be incorporated into it.
  • the invention comprehends a telephone set incorporating a subscriber interface unit as described herein for routing calls selectively via either the PSTN or a data network such as the Internet.
  • embodiments ofthe present invention provide selective routing of conventional POTS telephone communications via either the PSTN or a public data network without user intervention, facilitating optimization of route selection and transmission over the data network. • This may reduce the cost of communication and avoid the need for a PBX- to-data network switching device,
  • Embodiments of the invention advantageously enable standard telephone sets to communicate with other standard telephone sets by re-routing the communication through a public switched telephone system (PSTN) on to the public data system (Internet) and back onto the public switched telephone system (PSTN), thereby overcoming the requirement for access ' to a private data network of previous systems.
  • PSTN public switched telephone system
  • Internet public data system
  • PSTN public switched telephone system
  • IP data network
  • both broadband and dial-up embodiments o the invention advantageously allow deployment of scalable and cost-effective VoIP services without requiring (i) high density of subscribers in each district; (ii) permits and license fees from telephone companies; and (iii) purchase of expensive Central Office equipment which is likely to be underutilized as a result of low-density of subscribers requesting VoIP.
  • embodiments support normal emergency calls (police, ambulance) using regular PSTN connections during power outages.
  • the subscriber interface unit is easy to install ("plug and play"), and does not require any manual configuration from the end user.
  • the downloadable network selection rules allow for flexible subscription plans on a per customer basis, which allows the system to support only those VoIP calls that provide a competitive price advantage, for example "all long distance calls but not international calls”.
  • broadband embodiments of the invention effectively provide the end-user with an additional phone line, which could be used by another telephone set in parallel with the PSTN line. Because neither the user nor the user's subscriber premises interface unit (SPIU) needs to have any knowledge of the recipient's IP address, connectivity problems related to changes in the network configuration are avoided.
  • SPIU subscriber premises interface unit

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  • Engineering & Computer Science (AREA)
  • General Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Telephonic Communication Services (AREA)

Abstract

Selon l'invention, dans un procédé d'acheminement d'appels dans un système de télécommunications comprenant un RTPC (16) et un réseau public de données, une unité à interface d'abonné (12) produit la tonalité d'envoi pour un dispositif téléphonique standard (10), recueille les chiffres qui y ont été composés, en compare au moins quelques uns avec une gamme de numéros de destination pour déterminer s'il y a lieu ou non, selon des règles prédéterminées, d'acheminer l'appel sur le réseau de données ; se connecte par le biais du RTPC à un point de connexion (26) du réseau de données, transforme la communication d'une forme analogique en une forme numérique compressée, place la communication numérique sous forme VoIP, et achemine l'appel VoIP par le biais du réseau de données au point de sortie de connexion (30) de données qui convient à l'appelé. L'unité à interface d'abonné transforme sous forme analogique les signaux de données VoIP reçus de l'appelé par le biais du réseau de données et les communique au dispositif téléphonique de l'abonné.
PCT/CA2004/000396 2003-03-21 2004-03-19 Systeme et procede d'acheminement de communications telephoniques d'un telephone classique par le biais d'un reseau de donnees WO2004102941A1 (fr)

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US60/456,227 2003-03-21

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EP1770967A1 (fr) * 2005-10-03 2007-04-04 Lagunawave, Inc. Système voix sur IP (VoIP) avec acces d'appel local
WO2010009446A1 (fr) * 2008-07-17 2010-01-21 T-Mobile Usa, Inc. Système et procédé pour fournir de manière sélective des services de télécommunications entre un point d'accès et un réseau de télécommunication à l’aide d’un identifiant d'abonné
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US8619545B2 (en) 2008-07-17 2013-12-31 T-Mobile Usa, Inc. System and method for selectively provisioning telecommunications services between an access point and a telecommunications network based on landline telephone detection
US8774148B2 (en) 2009-02-27 2014-07-08 T-Mobile Usa, Inc. System and method for provisioning telecommunications services between an access point and a telecommunications network and providing missing information notification
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GB2414893B (en) * 2003-04-03 2006-12-13 Oki Electric Ind Co Ltd Voice communication apparatus
US7848495B2 (en) 2003-04-03 2010-12-07 Oki Electric Industry Co., Ltd. Voice communication apparatus
EP1770967A1 (fr) * 2005-10-03 2007-04-04 Lagunawave, Inc. Système voix sur IP (VoIP) avec acces d'appel local
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US9843480B2 (en) 2006-10-23 2017-12-12 T-Mobile Usa, Inc. System and method for managing access point functionality and configuration
US9301155B2 (en) 2006-10-23 2016-03-29 T-Mobile Usa, Inc. System and method for managing access point functionality and configuration
US8885635B2 (en) 2008-07-17 2014-11-11 T-Mobile Usa, Inc. System and method for selectively provisioning telecommunications services between an access point and a telecommunications network using a subscriber identifier
US8619545B2 (en) 2008-07-17 2013-12-31 T-Mobile Usa, Inc. System and method for selectively provisioning telecommunications services between an access point and a telecommunications network based on landline telephone detection
US8953620B2 (en) 2008-07-17 2015-02-10 T-Mobile Usa, Inc. System and method for selectively provisioning telecommunications services between an access point and a telecommunications network using a subscriber identifier
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US10880721B2 (en) 2008-07-28 2020-12-29 Voip-Pal.Com, Inc. Mobile gateway
US8774148B2 (en) 2009-02-27 2014-07-08 T-Mobile Usa, Inc. System and method for provisioning telecommunications services between an access point and a telecommunications network and providing missing information notification
US8189548B2 (en) 2009-03-06 2012-05-29 T-Mobile Usa, Inc. Authorizing access to telecommunications networks for mobile devices, such as mobile devices accessing networks via non-traditional entry points
US8484457B2 (en) 2009-03-10 2013-07-09 T-Mobile Usa, Inc. Method of securely pairing devices with an access point for an IP-based wireless network
US10932317B2 (en) 2009-09-17 2021-02-23 VolP-Pal.com, Inc. Uninterrupted transmission of internet protocol transmissions during endpoint changes

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