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WO2009001329A2 - Commutateur de téléphonie convergente de messagerie instantanée - Google Patents

Commutateur de téléphonie convergente de messagerie instantanée Download PDF

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Publication number
WO2009001329A2
WO2009001329A2 PCT/IL2007/000788 IL2007000788W WO2009001329A2 WO 2009001329 A2 WO2009001329 A2 WO 2009001329A2 IL 2007000788 W IL2007000788 W IL 2007000788W WO 2009001329 A2 WO2009001329 A2 WO 2009001329A2
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WO
WIPO (PCT)
Prior art keywords
skype
protocol
cts
service
call
Prior art date
Legal status (The legal status 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 status listed.)
Ceased
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PCT/IL2007/000788
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English (en)
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WO2009001329A3 (fr
Inventor
Arnon Azaria
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Individual
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Individual
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Priority to PCT/IL2007/000788 priority Critical patent/WO2009001329A2/fr
Publication of WO2009001329A2 publication Critical patent/WO2009001329A2/fr
Publication of WO2009001329A3 publication Critical patent/WO2009001329A3/fr
Anticipated expiration legal-status Critical
Ceased legal-status Critical Current

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Classifications

    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M7/00—Arrangements for interconnection between switching centres
    • H04M7/12—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal
    • H04M7/1205—Arrangements for interconnection between switching centres for working between exchanges having different types of switching equipment, e.g. power-driven and step by step or decimal and non-decimal where the types of switching equipement comprises PSTN/ISDN equipment and switching equipment of networks other than PSTN/ISDN, e.g. Internet Protocol networks
    • H04M7/1225—Details of core network interconnection arrangements
    • H04M7/123—Details of core network interconnection arrangements where the packet-switched network is an Internet Protocol Multimedia System-type network
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L65/00—Network arrangements, protocols or services for supporting real-time applications in data packet communication
    • H04L65/10—Architectures or entities
    • H04L65/102—Gateways
    • H04L65/1023—Media gateways
    • H04L65/1026—Media gateways at the edge
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L51/00—User-to-user messaging in packet-switching networks, transmitted according to store-and-forward or real-time protocols, e.g. e-mail
    • H04L51/04—Real-time or near real-time messaging, e.g. instant messaging [IM]

Definitions

  • the present invention generally relates to telecommunications services and to cross platform telecommunications interoperability (e.g. Telephony Switches and Gateway Protocols). More specifically, the present invention relates to "Interactive-Media Over Internet-Protocol” (IMOIP) enabled systems (sometimes called “Rich Media”, “Live Media”, or “Instant Messaging Broadband”) and methods that are preferably capable of providing "Private Branch Exchange” (PBX) services to the "Voice Over Internet Protocol” (VOIP) users, and furthermore to facilitate broader gateway interfaces between an Internet resident virtual PBX and other telecommunications infrastructures; such as legacy systems and/or mobile telephony service provider systems.
  • IMOIP Interactive-Media Over Internet-Protocol
  • PBX Primaryvate Branch Exchange
  • VOIP Voice Over Internet Protocol
  • PBX private branch exchange
  • the PBX allows calls (A) from many different outside parties to be directed via the PBX (with or without operator assistance) to various representatives within the enterprise, (B) between the representatives as intra-PBX conversations and in the form of call forwarding of an exterior caller from on representative to another; and from representatives to external "callers".
  • the standard POTS PBX provides (D) for automatic rotation of incoming calls to representatives, statistics about the performance of the representatives, voice or keypad navigation within the enterprise, message recording and message playing services, recordkeeping, cost allocation, and the likes.
  • VOIP PBX emulator to facilitate energy efficient telecommuting by representatives.
  • VOIP enabled QSIG like protocol to interconnect a plurality of virtual VOIP PBXs, since the VOIP enabled representatives (unlike traditional POTS enabled representatives) are scattered among various servers and physically reside in any of the worlds time zones.
  • the problem is to provide a complete package of PBX services to the VOIP client, without requiring the client neither to install nor to modify existing infrastructure; while facilitating hosted account type call center services with appropriate scalability.
  • many systems have been proposed and/or developed to try to address this critical aspect; including those described in US60239257 US20020118671 US20050047571 US20050282543 US20060023861 US20060193301 US20060239249 and US20060259668 (all of which are hereby incorporated by reference, as are all of the prior art references respectively cited therein). Nevertheless, none of these systems adequately facilitates a compact call center convergence between PBX and IM (Instant Messaging) VOIP sides.
  • IM-CTS Instant Messaging - Convergence Telephony Switch
  • the instant IM-CTS method, apparatus, and protocol are especially useful in man-computer interactions wherein there exists (in general) a need to provide a complete package of PBX services to the VOIP client, without requiring the client neither to install nor to modify existing infrastructure; while facilitating hosted account type call center services with appropriate scalability; and (in specific) to at least provide outgoing calling services from a caller of a VOIP hosted account bundle (virtual PBX) to a non-VOIP destination, such as a legacy device (POTS) or to a mobile telephone device.
  • POTS legacy device
  • an IM- Convergence Telephony Switch comprising: an internet resident software cluster 1301 having distributed therein a respective set of computer executable instructions (A) facilitating 1302 telephony call switching for an individual IM member among threads of a proxy hosted account, of multiple IM members within the same account, and the switching is between (I) at least one internet connected VOIP enabled communications device 1304 respectively using a first protocol and (II) at least one trunk 1305 of lines linked to an external Telephony Switch 1306 using a second protocol, and (B) facilitating 1303 respective telephony call processing for each thread when the thread is active, wherein said processing includes respective bidirectional transformations between a format of the first protocol and a format of the second protocol; wherein the first protocol is substantially an IM protocol and wherein the second protocol is substantially a protocol selected from the list: VOIP, Cellular, Digital, Analog, a combination of the at least two of the afor
  • the facilitating telephony call switching among individual threads of a proxy generally relates to service by an IM service provider for a customer, which in turn means action(s) on behalf of someone (i.e. surrogate). Therefore the IM service customer account is not resident on the customer hardware but is resident on the service provider machine and is managed by the service provider for the customer. Furthermore, a "hosted account" is of multiple IM members within the same account; and therefore comes to utilize a plurality of threads. Likewise, it should be noted that when dealing with a PBX, trunk lines are the phone lines coming into the PBX from the telephone provider.
  • Trunk lines transmit voice and/or data in formats such as analog, Tl, El, ISDN or PRI, VOIP (SIP, H323, MGCP, SS7), cellular and Wireless. Trunks can use protocols that connect via: Copper, Internet, cellular, wireles, etc. (mutatis mutandis).
  • VOIP Voice over IP
  • MGCP Mobility Management Protocol
  • SS7 Temporal Transport Protocol
  • RVP Remote Access Protocol
  • SAPv2 Session Control Protocol
  • SGCP Session Control Protocol
  • Skinny Session Control Protocol
  • TAPI JTAPI
  • RTSP any earlier version of any of the aforesaid, or the likes.
  • a Cellular protocol is GPRS, GSM, SS7, AMPS, CDMA, CDMA2000, CSD, DataTac, DECT, EDGE, EVDO, FDMA, UMA, GAN, HCSD, HSDPA, iDEN, Mobitex, NMT, PDC, PHS, TACS, TDMA, TD- SCDMA, UMTS, WCDMA, WiDEN, any earlier version of any of the aforesaid, or the likes.
  • a Digital protocol is ISDN, PRI, El, Tl, TDM, BRI, BRITE, any earlier version of any of the aforesaid, or the likes.
  • an Analog protocol is selected from the list: FXS, FXO, any earlier version of any of the aforesaid, or the likes.
  • the external Telephony Switch is a specific PBX, a telephony operator switching system, a cellular switch, or the likes.
  • the trunk of lines is selected according to a conditional structure in the list: (A) for a VOIP environment, the trunk is an integer multiple of 8 lines; (B) for an ISDN El environment, the trunk is an integer multiple of 31 lines; (C) for an ISDN Tl environment, the trunk is a multiple of 24 lines; (D) for analog FXO FXS environment, the trunk is an even integer number of lines; and (E) for a cellular environment, the trunk is an even integer number of lines.
  • a communications device of the at least one communications device is enabled to access the internet via a connection medium selected from the list: Wifi, ADSL, Wireless, Wire line, Cellular, and the likes.
  • the substantially IM protocol is Skype. Please note that most of the detailed examples presented herein are in terms of Skype, however intrinsically one may substitute the detailed implementations via Skype with equivalent detailed implementations via other IM-Service "provider" systems (e.g. the substantially IM protocol is selected from the list: a Skype protocol, a Lycos Phone protocol, a Microsoft Messenger protocol, a Yahoo Messenger protocol, a Google Talk protocol, or the likes).
  • the proxy hosted account is associated with at least one IM-Service account ID (respective IM-Service Name) selected from the IM- Service providers Skype, Lycos Phone, Microsoft Messenger, Yahoo Messenger, and Google Talk.
  • the proxy hosted account is associated with at least one IM-Service account ID (respective IM-Service Name).
  • facilitating telephony call switching includes associating the at least one IM-Service account ID with at least one respective IM-Service extension or with a plurality of concurrent respective IM-Service extensions.
  • facilitating telephony call switching includes software that, for an incoming call from a VOIP enabled device to a IM-Service account ID having the at least one IM-Service extension, firstly facilitates identifying available IM-Service extensions of the IM- Service account ID, secondly facilitates connecting the incoming call to one of the identified available IM-Service extensions, thirdly facilitates call switching between (I) an available IM- Service extension of the identified available IM-Service extensions and (II) the at least one trunk, and fourthly facilitates propagating the incoming call to a line of the at least one trunk.
  • the set of computer executable instructions includes software that, for an incoming call originating from an external Telephony Switch via an attached trunk, firstly facilitates identifying the IM- Service account ID associated with the trunk, secondly facilitates identifying available IM- Service extensions of the IM-Service account ID, thirdly facilitates connecting the incoming call to one of the identified available IM-Service extensions, and fourthly facilitates propagating the call to the destination VOIP communications device.
  • facilitating telephony call switching includes facilitating identifying the individual IM member, among the multiple IM members within the same account, by performing an automatic activity selected from the list: a directory search, a table lookup, a telephone "number” transformation, and an IM "calling code” concatenation; wherein the call switching is for a call originating from a cellular telephone or from a landline telephone, and is propagating to the VOIP communications device.
  • the internet resident software includes at least one set of computer executable instructions residing on a hardware device selected from the list: a server, a cell phone, a wireless communications enabled device, and a computer apparatus.
  • a hardware device selected from the list: a server, a cell phone, a wireless communications enabled device, and a computer apparatus.
  • IM-CTSs mutually interconnected to the internet allow for the instant configure of a new class of global telephone enterprises.
  • placing micro-IM-CTSs on respective cellular telephones and/or on respective telephone-and-internet systems connected computer apparatus (PCs, etc.) allows for a peer-to-peer style scalable global telephone cooperative; potentially accommodating disparity of external billing events to respective local terrestrial and/or POTS telephone companies to financial load balancing models known from other non-telecommunications cooperatives.
  • an IM- Convergence Telephony Switch substantially relates to a set of computer executable instructions 1101 (or 1201) for managing call throughput connections including software that, for an incoming call 1102 to a "hosted account” 1103, firstly facilitates 1202 identifying available VOIP extensions 1104 associated with at least one respective "hosted account” and secondly facilitates 1203 connecting the incoming call to one of the identified available VOIP extensions 1105.
  • This embodiment focuses on the call center routing facility of a basic PBX implemented for an internet centric telephone instantiation; preferably with interconnection to at least one external telephony switch.
  • Most progressive VOIP implementations are designed to facilitate rich media intercommunications; the substantially simultaneous transmission and receipt of high quality streaming audio, streaming visual, and text parcels (e.g. documents, IM-CTS, SMSs, etc.). Rich media interconnection runs smoothly if the interconnected users have broadband access.
  • the rich media VOIP service facilitator does not really care about the wasted bandwidth of users who use this connection for voice and/or text without streaming video or recording studio quality music.
  • the rich media VOIP service facilitator is focused on maintaining highest quality of service, even if there is lots of allocated albeit unused bandwidth and/or lots of empty packets running back and forth.
  • a software application for the ordinary low- to-intermediate user group e.g. call center representatives
  • a front-end interface that organizes traffic between the representatives, through the physical bandwidth limited internet access interconnect, and to/from software localized (or clustered) in the internet.
  • routing between the localized or clustered software ends may be accomplished using the broadband VOIP service facilitator, over private networks, via commercial telecommunications carriers, or the likes.
  • the rich media VOIP service facilitator may provide easy PBX services to the users, in that a single user often keeps an instantiation of his user-to-VOIP application on multiple computers (e.g. on his work computer, his home computer, etc.).
  • the VOIP provider wants to signal the user that he has an incoming call, typically all of the users running instantiations will ring.
  • the provider will automatically cancel and temporarily suspend interactions with the other instantiations - until the current accepted call in terminated.
  • the rich media VOIP service facilitator may provide easy PBX services to the users, in that dialing-codes, area-codes, and country-codes are part of accepted legacy telephony practice.
  • a unique access code or "termination" for each VOIP service provider; thereby facilitating instant access among VOIP service facilitators, and between respective VOIP service facilitators and/or respective Legacy (POTS) service providers and/or mobile telephone service providers, WIFI, etc.
  • POTS Legacy
  • a basic embodiment of the instant invention relates in general to embodiments hosting Skype accounts and routing converging of Skype calls between Skype users and legacy telecommunication switches belonging to enterprises, PSTN operators, and cellular operators; and preferably vice versa. More specifically, a basic embodiment of the instant invention relates to a method (logical apparatus - i.e. software) for routing and converging calls between Skype users and legacy telecommunication switches; by using a IM-CTS (Skype), (the IM-CTS (Skype) is an implementation example of Skype services; much as a Google Talk Gateway Switch would be an IM-CTS implementation example of Google Talk Service, etc.) which facilitate Skype accounts hosting for enterprises and telephone companies, and enables call convergence and routing between these organizations and Skype users.
  • IM-CTS IM-CTS
  • IM-Convergence Telephony Switch IM- CTS
  • FXS Foreign Exchange Station
  • TDM Time-Division Multiplexing
  • SIP Session Initiation Protocol
  • MGCP Media Gateway Control Protocol
  • SS7 Signaling System 7
  • Skype telephony has gained momentum as a leading Internet telephony service, which support a diverse means of end user devices for communications.
  • Skype users can communicate by using their personal computer (PC), PC attached phones, special PSTN phones, and WLAN phones.
  • Skype users can benefit by using the Skype-In services. This free easy-to-use service has gain momentum and popularity among home users.
  • the benefits of communications between Skype subscribers can be extended to organizations such as enterprises and telecommunications service providers. These organizations have invested, throughout the years, a great deal of resources in legacy communications equipment; such as Public Branch Exchange (PBXs), PSTN Switches, and Cellular Switches. To enable these organization to join the Skype network, their legacy telecommunications equipment can be converged to the Skype network, using an embodiment of the instant invention, and interoperate with the offered Skype services.
  • PBXs Public Branch Exchange
  • PSTN Switches PSTN Switches
  • Cellular Switches Cellular Switches
  • embodiments of the instant invention relate to a method and apparatus for facilitating session control in an inter-protocol convergence between the Skype protocol and the telecommunication equipment protocols, which may include the following telecommunications protocols: TDM, FXS, FXO, SS7, SIP 5 H323, MGCP and others.
  • IM-CTS Skype enabled ATC embodiment, "proxy" according to the instant invention.
  • the IM-CTS (Skype) hosts Skype accounts for these organizations. Any communication that is directed by Skype to these organization's Skype accounts, is converged by the IM-CTS (Skype) to the target telephony protocol and forwarded to the organization legacy telecommunication equipment.
  • the IM-CTS serves as an intermediary between the Skype network and the telecommunication legacy equipments.
  • a call from a e ⁇ urce Skype subscriber to a destination organization Skype account would be connected via the Internet to the organization Skype account hosted at the IM-CTS (Skype), which in turn converges the Skype protocol to a the target protocol (i.e. SIP, TDM 5 SS7, H323 and others), and than forward the call the Internet to the destination telecommunication equipments.
  • a target protocol i.e. SIP, TDM 5 SS7, H323 and others
  • an instant invention embodiment narrow instantiation relates to "Inter-site call routing and roaming support"; particularly to a method and apparatus, arranged and operating for hosting Skype accounts at the IM-CTS (Skype).
  • the hosted Skype accounts within the IM-CTS (Skype) (100) may belong to an organization such as enterprises 150, or telecommunication service providers such as PSTN (Public Switched Telephone Network) 16O 5 or to Cellular Operators 170.
  • PSTN Public Switched Telephone Network
  • a call from a source Skype subscriber (110, 120, or 130) is received at the IM-CTS (Skype) (100) via the Internet, it converges the Skype telephony protocol to the destination telephony protocols (e.g. SIP) 5 and forwards the call via the Internet 140 to destination organization (150, 160, or 170) telephony switch for call completion.
  • a typical Skype IM-CTS is a Virtual Internet Skype-based convergence PBX wherein: (A) the IM-CTS system can host on one hand Skype-based services for Enterprises and Telco's, and on another hand SIP-based services, and connect the two together; and in particular a Layer 2 within the IM-CTS subsystem enables a unique convergence protocol between the SIP-based subsystem and the Skype-based subsystem. It uses an implementation version of the instant invention "IM-CTS Codec" (VOIP conversion) that is described below.
  • IM-CTS Codec VOIP conversion
  • Skype wideband voice format to a narrowband VOIP format (e.g. SIP)
  • VOIP format e.g. SIP
  • a major feature in the IM-CTS system is its ability to enable multiple concurrent Skype phone calls (Skype extensions) for every single ordinary Skype account ID (can be an enterprise Skype account ID).
  • a typical IM-CTS customer can use the system in the following manner: a. Define a Skype account ID via the standard Skype Web facilities. The Skype account is a standard Skype account. b. Once the Skype Account is defined, the IM-CTS customers register theirs Skype account within the IM-CTS online Web registration facilities. c. Once defined to the IM-CTS system, the customer can select the number of Skype extensions that can be activated concurrently under his/her Skype account ID. d. IM-CTS enables to customer to select unlimited number extensions per single Skype Account ID. e. The Skype extensions that can be activated under the same single Skype account ID, are organized in trunks:
  • trunks as multiple of 8, i.e.: 8, 16, 32, 64, 128....
  • trunks as multiple of El and Tl lines, i.e.: 1 El line is 31 lines and 1 Tl line is 24 lines.
  • the IM-CTS system is enabling multiple Skype extensions under single Skype Account as described below.
  • the IM-CTS is enabled to run multiple concurrent Skype telephone sessions under a single Skype account Id by using the following methodology:
  • Each IM-CTS server consists of the (I) The IM-CTS Manager - this is the IM- CTS calls trafficker; and (II) Skype extensions each running under a separate Linux account- ID. Each Skype account is attributed a specified number of concurrent Skype extensions. It is implemented in a single Linux Server within the IM-CTS environment.
  • FIG. 8 schematically illustrates that each Skype extension within the trunk for customer A (804) and B (806) is: Running under a separate Skype user-ID (under the same IM-CTS); Activated by a standard Skype-supplied client; and the Skype client is activated with the Skype ID that has been defined and select by the customers (A or B).
  • the IM-CTS manager o Overlooks at all telephony activities within the IM-CTS environment o Incoming call (802) to a IM-CTS hosted Skype account (804) proxy: Identify the available Skype extensions (805) for the specific Skype account (804); connects the call to extension o Outgoing call (807) from a IM-CTS hosted account proxy: Identify the destination (a specified PBX or telephony operator switching system), and propagate the call to the destination. o Enforce security and integrity of the IM-CTS system
  • Customer A hosted account (804) has selected to have 8 concurrent Skype extensions hosted at the IM-CTS system, and propagated to his SIP PBX as a single trunk with 12 SIP cannels.
  • Fig. 9. illustrates a complete call-flow from an incoming source PBX (903) to an outgoing destination Skype subscriber (907):
  • An example phone call from a SIP source PBX subscriber (903) is initiated to a destination Skype subscriber (907), such that — (FIRST)
  • a phone call is initiated by a SIP user within a specified PBX, to a destination Skype subscriber, by calling the Skype hosted account (904) which is hosted within the IM-CTS system (901), as are other Skype hosted accounts such as 906.
  • the IM-CTS Manager (902) receives the call signaling, and identifies the Skype hosted account (904) within the IM-CTS (901).
  • IM-CTS Manager The IM-CTS Manager (902), checks the availability and validity of the Skype extension (905), and_connects the call to it. Once connected, the Skype extension (905) propagates the call to the outgoing destination Skype subscriber (907), and the call is established.
  • Fig. 9 can also be used to illustrate the reverse a call flow - when the source incoming call is from a VOIP enabled device (903) to a destination telephone which is attached to a PBX.(907) : A phone call from a VoIP enabled device (903) to a hosted Skype account (904) to a destination SIP source PBX (907).
  • Skype user such that (First) A phone call is initiated by a VOIP enabled device (903) to a hosted Skype account (904) within the IM-CTS 5 to a destination SIP PBX (907).
  • the IM-CTS Manager (902) receives the call signaling, and identifies the Skype hosted account (904) and the associated Skype extensions (905), connects the call to the Skype extension (905).
  • the Skype extension (905) propagates the call to the target SIP PBX system (907).
  • the IM-CTS Manager (902) checks the availability and validity of the SIP trunk which leads to the SIP PBX (907), and propagates the call to it. Once handshaking is completed, the call is established.
  • IM-CTS IM-CTS Adaptive Mechanism
  • An object of the IM-CTS Adaptive Mechanism is to enable the operation of the IM-CTS IM-Convergence telephony switch (IM-CTS) in a constantly changing environment.
  • SAM IM-CTS IM-Convergence telephony switch
  • Skype wideband bandwidth usage is 3-16 kilobytes/second and preferred IM-CTS telephony switch bandwidth usage is 16 kilobits/second.
  • Sampling frequency Skype sampling rate is variable between 3ms to 10ms (330 Hz to 100 Hz, respectively) and typical Legacy telephone systems sampling rate is 100ms (10 Hz).
  • Skype sampling rate is variable between 3ms to 10ms (330 Hz to 100 Hz, respectively) and typical Legacy telephone systems sampling rate is 100ms (10 Hz).
  • Multi-User Interference (MUI) An embodiment of the instant method exhibits adaptive capabilities with respect to MUI degrading effects and, depending on number of Skype extensions for a single Skype account-ID, the IM-CTS system perform an adaptive prediction to determine and make a decision as for "when to issue a buffer-read request" from communicating partner with minimal impact of MUI on sampling rate frequency.
  • Each Skype subscriber may be located in a different geographical location, have a different Internet connectivity (modem type, wireless, etc.) and bandwidth. This above constrains cause differences in the VOIP sampling rate, as one Skype account may include multiple Skype extensions which are exchanging VOIP buffers with different Skype subscribers.
  • Dynamic Jitter In order to compensate the differences of sampling rates frequencies for the Skype wide-band and the legacy telephone narrow-band, a dynamic jitter buffer mechanism is implemented.
  • IM-CTS Adaptive Mechanism (SAM) operation For each telephony session between IM-CTS Skype-extension (which belongs to a hosted Skype account-ID) and another Skype subscriber, SAM collects the following four information items: (I) Number of active telephony sessions under the Skype account-ID. (II) Number of active telephony sessions in the server. (Ill) Time delay for a complete cycle: starting at issuing "buffer read request" (from communicating Skype peer), going thru receiving the buffer, down-sampling, finishing by accumulating in jitter buffer.(IV) Network time delay - how long it takes the communicating Skype peer to return a buffer after a "buffer read request". These four information items are classified by the Skype ID of the Skype party which is communicating with the IM-CTS system. The information is kept in a database, which resides at the IM-CTS system.
  • SAM When an active telephony session is started SAM is initiating a dynamic jitter buffer that mediate the gap of sampling rate frequency between Skype and the legacy telephony equipment. For each active session, SAM builds a prediction model which calculates the exact time to issue the next "buffer read request". SAM applies statistical analysis that takes in count the following dimensions: (I) Historical Analysis: Associated with the specific Skype communicating party and Associated with the IM-CTS "hosted account" (proxy) system under similar load and constrains. (II) Real-time analysis: Network load; Multi User Interference (MUI); and End-to-end response (include network response and processing). (Ill) SAM analysis determines what should be the time delay until issuing the next "buffer read request”.
  • MUI Multi User Interference
  • End-to-end response include network response and processing.
  • buffer read request (I) The VOIP buffer from communicating Skype peer transmitted to the IM-CTS system. (II) Down-sampling the buffer (III) Accumulating the buffer within the dynamic jitter buffer (IV) Every 100ms (10 Hz), SAM takes 1600 bits of data (represents an accumulation of 100ms, in a 16 kilobits/second codec), and transmits it to the legacy telephony system.
  • FIG. 1 is a diagram illustrating a simplified and representative environment associated with an exemplary architecture of the IM-CTS (Skype) (100) having enterprise sites with exemplary PBXs (150), exemplary telecommunication service providers with exemplary telephony switch (160, 170), and exemplary Skype users (HO 5 120, 130) and connections to a Internet (140) in accordance with various exemplary embodiments;
  • IM-CTS IM-CTS
  • FIG. 1 is a diagram illustrating a simplified and representative environment associated with an exemplary architecture of the IM-CTS (Skype) (100) having enterprise sites with exemplary PBXs (150), exemplary telecommunication service providers with exemplary telephony switch (160, 170), and exemplary Skype users (HO 5 120, 130) and connections to a Internet (140) in accordance with various exemplary embodiments;
  • FIG. 2 is a diagram illustrating the exemplary Skype end user having an exemplary Personal Computer PC (210) with Skype services, or an exemplary Skype wireless telephone (221) connected to the Internet via a wireless router (220) , or an exemplary Skype-In PSTN phone number (230) which facilitate Skype communication with a PSTN subscriber, in accordance with various exemplary embodiments;
  • FIG. 3 is a diagram illustrating an exemplary enterprise having sites with exemplary IP PBX (310) and IP link to the Internet or exemplary TDM PBX (321) connected to the Internet via a TDM Gateway (320) in accordance with various exemplary embodiments;
  • FIG. 4 is a diagram illustrating an exemplary PSTN Operator having sites with exemplary IP PSTN Switch (400) and IP link to the Internet or exemplary TDM or SS7switch (411) connected to the Internet via TDM or SS7 gateway 410 in accordance with various exemplary embodiments;
  • FIG. 5 is a diagram illustrating an exemplary Cellular Operator having sites with exemplary SS7 (500) switch and IP link to the Internet or exemplary SS7 switch connected to the Internet via a SS7 gateway (510) in accordance with various exemplary embodiments;
  • FIG. 6 is a block diagram illustrating exemplary call flow between an exemplary source Skype User (600) to an exemplary destination enterprise or telecommunication service provider (620) via the exemplary IM-CTS (Skype) 610 in accordance with various exemplary embodiments;
  • FIG. 7. is a diagram illustrating an exemplary apparatus in accordance with various exemplary and alternative exemplary embodiments.
  • FIG. 8 illustrates the IM-CTS (801) structure, where for each customer A (804) and B (806) is: Running under a separate Skype user-ID (804) (under the same IM-CTS); Activated by a standard Skype-supplied client (805) for connecting incoming call (802) to an outgoing call (807);
  • Fig. 9. illustrates a complete call-flow from an incoming source PBX (903) to an outgoing destination Skype subscriber (907): and vice- versa - a call flow when the source incoming call is from a VOIP enabled device (903) to a destination telephone which is attached to a PBX.(907) ;
  • Fig.10 is a schematic diagram of an IM-Convergence Telephony Switch (IM-CTS) network topology
  • Fig.11 is a schematic diagram of an IM-Convergence Telephony Switch (IM-CTS) structure
  • Figs 12-13 are schematic diagrams illustrating two forms of exemplary IM-CTS.
  • the present invention concerns communications between Skype subscribers and organizations such as enterprises and telecommunication service providers via the IM-CTS (Skype).
  • IM-CTS telecommunication service providers
  • the wireless units such as two-way radios and the like associated with a communication system such as a home network or enterprise network, a WLAN or the like.
  • IM-CTS IM-CTS
  • methods therein for facilitating Skype communication to organization such as enterprises or telecommunication service providers by receiving the Skype call at the IM-CTS (Skype), which in turn converges the Skype protocol to a target protocol such as TDM, SIP, H323, SS7 and others, and establishes connectivity to the target Enterprises PBX or the telecommunications service providers switches via the Internet.
  • target protocol such as TDM, SIP, H323, SS7 and others
  • the IM-CTS (Skype) (100) serves as an intermediate communication switch which hosts Skype accounts for organizations. It receives calls from Skype users 110-130 with the Skype protocol, converge the calls to a destination protocol and foreword the call to the destination enterprise or destination telecommunication service provider (150-170). Calls that are initiated by enterprise phone subscriber, via the PBX or by PSTN phone subscriber, are forwarded by the PBX or PSTN switch to the IM-CTS (Skype) (100), that converges the call to the Skype protocol and forewords the call via the Internet to the Skype users (110-130).
  • the IM-CTS converges the Skype calls to voice communications protocols such as the TDM protocol, VOIP protocols such as SIP, H323 or MGCP , various cellular phone protocols such as, analog and digital cellular, CDMA (code division multiple access) and variants thereof, SCCAN, GSM, GPRS (General Packet Radio System), 2.5 G and 3 G systems such as UMTS (Universal Mobile Telecommunication Service) systems, 3GPP, 3GPP2, 4G, PTT, Internet Protocol (IP) Wireless Wide Area Networks like 802.16, 802.20 or FLASH-Orthogonal Frequency Division Multiplexing (OFDM) network, integrated digital enhanced networks and variants or evolutions thereof.
  • voice communications protocols such as the TDM protocol, VOIP protocols such as SIP, H323 or MGCP , various cellular phone protocols such as, analog and digital cellular, CDMA (code division multiple access) and variants thereof, SCCAN, GSM, GPRS (General Packet Radio System), 2.5 G and 3 G systems such as
  • FIG. 1 A simplified and representative exemplary scenario associated with an exemplary configuration is illustrated in FIG. 1.
  • the Skype services can be obtained by various end user devices such as Personal Computer (PC) equipped with the Skype software (110), Skype software installed on a wireless phone in WLAN at home or enterprises or public places (130), or Skype-IN PSTN phone number (120) that is routed by Skype to a particular Skype user (120).
  • An exemplary target organization may be an enterprise (150), or telecommunication service providers such as PSTN (Public Switched Telephone Network) (160) or LEC (Local Exchange Carrier) or IXC (Inter-exchange carrier).
  • PSTN Public Switched Telephone Network
  • LEC Land Exchange Carrier
  • IXC Inter-exchange carrier
  • the IM-CTS (Skype) (100) operates as convergence and forwarding protocol between the calling parties. It receives phone calls from the Skype users (110-130), converges the calls to a target protocols, and foreword the calls to the target communication switch (150-170).
  • Skype services can be provided by various end-user devices such as Personal Computer (PC) equipped with the Skype software (210), Skype software installed on a wireless phone in WLAN at home or enterprises or public places (221) and utilizes a wireless router (220) to access the Internet, or Skype-IN PSTN phone number (231) that is routed by Skype to a particular Skype user (230).
  • PC Personal Computer
  • Skype software installed on a wireless phone in WLAN at home or enterprises or public places (221) and utilizes a wireless router (220) to access the Internet, or Skype-IN PSTN phone number (231) that is routed by Skype to a particular Skype user (230).
  • An exemplary target organization may be an enterprise which utilizes an IP PBX (310), with PBX attached phones (311) or an enterprise which utilizes a TDM PBX (321) with TDM gateway (320) to facilitate VOIP communications.
  • Fig 4. exemplifies a telecommunication service providers such as PSTN (Public Switched Telephone Network) or LEC (Local Exchange Carrier) or IXC (Inter-exchange carrier) which utilizes a VOIP switch (400) having a PSTN telephone (401) subscriber attached, or such an organization which utilizes a TDM switch (411) connected to a TDM to VOIP gateway (410).
  • Fig 5 exemplified a Cellular Operator with a VOIP switch 500 having a cellular telephone (501) subscriber in communications, or a Cellular Operator SS7 switch (510) which utilizes an SS7-VOIP Gateway (511).
  • an exemplary telecommunication switch in FIG 1. such as any one of PBX (150), telecom switch (160), and Cellular Switch (170), needs to know how to route a call to a destination telephone which are attached to it.
  • an exemplary destination Telecom Switch (160) should have the information to which attached telephone the call should be routed.
  • all target PBXs and switches can be configured with domain identifiers in the form of a directory search, a table lookup, a telephone "number” transformation, and an IM "calling code” concatenation system that can further assist in identifying the target telephone.
  • an exemplary PSTN telephone which is connected via copper wires to a PSTN telecommunication switch such as any one of telecom switch (160), initiate a call to a target Skype subscriber by selecting a predefine Skype user with a predefined attached a directory search, a table lookup, a telephone "number” transformation, and an IM "calling code” concatenation.
  • the Skype user IDs and the associated speed dial which reside at the IM-CTS (Skype) (100)
  • IM-CTS (Skype) 100
  • FIG. 6 A more detailed example of the operation of a call process in accordance with various exemplary embodiments is shown in FIG. 6.
  • a call is initiated, for example, at Skype wireless phone or PC equipped with Skype software (600) to a destination Skype account which is hosted at the IM-CTS (Skype) (610).
  • Skype software 600
  • IM-CTS Sekype
  • the call is received at the IM-CTS (Skype) (610), the call is processed by the Skype Protocol Converter (611) and converted to the selected target telephony protocols, such as: VOIP protocol (612) as SIP, H323, and others; Legacy protocols (613) as TDM, FXS, FXO and others; Cellular protocols (614) as SS7 and others;
  • the call is processed by the IM-CTS manager (615) and transferred via telecommunication lines to the organization switch (620).
  • the organization switch (620) receives the call, and completes the call process by forwarding the call to the appropriate destination telephones, such as: for enterprise PBX a PBX phone, for PSTN switch a PSTN phone, and for cellular operator a cellular phone.
  • a call is initiated, for example, at PSTN phone or Cellular Phone, forwarded to PSTN Switch or Cellular switch (620), and than forwarded to the IM-CTS (Skype) (610).
  • the IM- CTS manager (615) accepts the call and process it with the appropriate protocol such as: VOIP protocol (612) as SIP, H323, and others; Legacy protocols (613) as TDM, FXS, FXO and others; Cellular protocols (614) as SS7 and others; Once converted, the call is processed by the Skype protocols converter (611) and transmitted over communication lines to the Telephone Destination (600) Skype user.
  • apparatus (700) may be an exemplary device such as an exemplary inter-domain telephony switch, a server computer, a cellular phone or the like without departing from the invention.
  • apparatus (700) preferably includes processor (710), memory (720), storage (730), network interface (750 and 760) may provide an interface to a network such as the Internet network, a private IP network, Local Area Network (LAN), a Public Switched Telephone Network (PSTN) or other network as would be appreciated by one of ordinary skill.
  • a network such as the Internet network, a private IP network, Local Area Network (LAN), a Public Switched Telephone Network (PSTN) or other network as would be appreciated by one of ordinary skill.
  • LAN Local Area Network
  • PSTN Public Switched Telephone Network
  • Apparatus (700) may also be provided with user interface (740), particularly when apparatus (700) is embodied as a user interface or the like, although in many instances no user interface will be necessary. It will further be appreciated that as the inventive principles described herein are suitable for implementation in, for example, a software program, the instructions associated with the computer program and capable of being read by processor (710), may preferably be stored in memory (720) and may be executed in order to perform the useful functions and routines described herein. Now, according to a narrow aspect of the instant invention, there is provided a method for hosting Skype accounts and routing and converging of Skype calls between Skype users and legacy telecommunication switches belonging to enterprises, PSTN operators and cellular operators, and vice versa.
  • IM-CTS IM-CTS
  • WAN Wide Area Network
  • a IM-CTS that converges the Skype telephony protocol to a plurality of telephony protocol; such as the TDM protocol, VOIP protocols such as SIP, H323 or MGCP , various cellular phone protocols such as, analog and digital cellular, CDMA (code division multiple access) and variants thereof, SCCAN, GSM, GPRS (General Packet Radio System), 2.5 G and 3 G systems such as UMTS (Universal Mobile Telecommunication Service) systems, 3GPP, 3GPP2, 4G, PTT, Internet Protocol (IP) Wireless Wide Area Networks like 802.16, 802.20 or FLASH-Orthogonal Frequency Division Multiplexing (OFDM) network, integrated digital enhanced networks and variants or evolutions thereof, and vice Vera.
  • IP Internet Protocol
  • IP Wireless Wide Area Networks like 802.16, 802.20 or FLASH-Orthogonal Frequency Division Multiplexing (OFDM) network, integrated digital enhanced networks and variants or evolutions thereof, and vice Vera.
  • IP Internet Protocol
  • the IM-CTS receives Skype calls to the plurality hosted Skype accounts and converges and foreword the Skype calls a destination telecommunication switches such as enterprise PBX, PSTN switches, and Cellular switches.
  • the hosted Skype accounts belong to organizations such as: enterprises, PSTN operators, cellular operators and any other telephony service provider.
  • the IM-CTS (Skype) receives Skype calls to the hosted Skype accounts, converges the Skype protocol to the destination telephony protocol and forward the call to the destination telephony switch.
  • the telephony protocol includes a Session Initiation Protocol (SIP).
  • SIP Session Initiation Protocol
  • the telephony protocol includes a H323; or a Media Gateway Control Protocol (MGCP); or a Time-Division Multiplexing (TDM); or an Integrated Service Digital Network (ISDN); or a Signaling System 7 (SS7); or a PSTN telephony switch; or an Internet Protocol (IP) compliant Private Branch Exchange (PBX); or a Cellular telephony switch; or the likes.
  • MGCP Media Gateway Control Protocol
  • TDM Time-Division Multiplexing
  • ISDN Integrated Service Digital Network
  • SS7 Signaling System 7
  • PSTN telephony switch or an Internet Protocol (IP) compliant Private Branch Exchange (PBX); or a Cellular telephony switch; or the likes.
  • IP Internet Protocol
  • PBX Private Branch Exchange
  • the IM-CTS receives telephone calls from telecommunication switches such as enterprise PBX 5 PSTN switches, and Cellular switches; in-turn convergence the calls to the Skype protocol and forward the calls to the Skype network via the hosted Skype accounts.
  • telecommunication switches such as enterprise PBX 5 PSTN switches, and Cellular switches
  • the IM-CTS resides at a plurality of telecommunication center premises which may include: enterprise communication facility, PSTN operator premises, Cellular operator premises or any other telecommunication facilities.
  • the IM-CTS (Skype) communicates over Wide Area Network (WAN) communication lines such as the public Internet network, Virtual Private Networks (VPN), private communication lines, and Local Area Network (LAN).
  • WAN Wide Area Network
  • VPN Virtual Private Networks
  • LAN Local Area Network
  • a method for facilitating inter-site call routing where sites may be connected through a network associated with the public Internet or private IP networks.

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  • Engineering & Computer Science (AREA)
  • Multimedia (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Telephonic Communication Services (AREA)
  • Data Exchanges In Wide-Area Networks (AREA)
  • Mobile Radio Communication Systems (AREA)

Abstract

La présente invention concerne un commutateur de téléphonie convergente de messagerie instantanée (IM-CTS) comportant : un groupe logiciel résident sur l'Internet comprenant des instructions informatiques permettant respectivement (A) de faciliter la commutation d'appels téléphoniques pour un abonné individuel de messagerie instantanée parmi des fils d'un compte de compte de serveur mandataire hébergé, d'une pluralité d'abonnés de messagerie instantanée à l'intérieur du même compte, et la commutation entre (I) au moins un dispositif de communications capable de voix sur IP connecté à l'Internet au moyen d'un protocole essentiellement de type messagerie instantanée et (II) au moins une jonction dont les lignes sont reliées à un commutateur de téléphonie externe sous un second protocole (par exemple, voix sur IP, cellulaire, numérique, analogique), et (B) de faciliter le traitement d'appels téléphoniques respectifs pour chaque fil lorsque le fil est actif. De préférence, le protocole essentiellement du type messagerie instantanée est du type Skype, Microsoft Messenger, Google Talk, ou analogues.
PCT/IL2007/000788 2007-06-28 2007-06-28 Commutateur de téléphonie convergente de messagerie instantanée Ceased WO2009001329A2 (fr)

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PCT/IL2007/000788 WO2009001329A2 (fr) 2007-06-28 2007-06-28 Commutateur de téléphonie convergente de messagerie instantanée

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Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP2668741A1 (fr) * 2011-07-27 2013-12-04 Vonage Network, LLC Systèmes et procédés de fourniture de services de communication
US20140032774A1 (en) * 2012-07-30 2014-01-30 Microsoft Corporation Client-emulating Gateways for Communication Network Migration
US9854102B2 (en) 2011-07-27 2017-12-26 Vonage America Inc. Systems and methods of providing communications services
EP2254319B1 (fr) * 2009-05-22 2018-07-11 Comcast Cable Communications, LLC Intégration de services de discussion vocale
US10973059B2 (en) 2011-07-27 2021-04-06 Vonage America, Llc Systems and methods of providing communications services

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* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US7313134B2 (en) * 2005-05-12 2007-12-25 Yahoo! Inc. Proxy server for relaying VOIP messages

Cited By (9)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
EP2254319B1 (fr) * 2009-05-22 2018-07-11 Comcast Cable Communications, LLC Intégration de services de discussion vocale
EP2668741A1 (fr) * 2011-07-27 2013-12-04 Vonage Network, LLC Systèmes et procédés de fourniture de services de communication
EP2668741A4 (fr) * 2011-07-27 2014-01-01 Vonage Network Llc Systèmes et procédés de fourniture de services de communication
US9854102B2 (en) 2011-07-27 2017-12-26 Vonage America Inc. Systems and methods of providing communications services
US10973059B2 (en) 2011-07-27 2021-04-06 Vonage America, Llc Systems and methods of providing communications services
US11503084B2 (en) 2011-07-27 2022-11-15 Vonage America Inc. Systems and methods of providing communications services
US20140032774A1 (en) * 2012-07-30 2014-01-30 Microsoft Corporation Client-emulating Gateways for Communication Network Migration
WO2014022250A1 (fr) * 2012-07-30 2014-02-06 Microsoft Corporation Passerelle d'émulation de clients pour la migration de réseaux de communication
CN104509070A (zh) * 2012-07-30 2015-04-08 微软公司 用于通信网络迁移的客户端模拟网关

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