EP1636977A2 - Convergence entre des communications vocales en mode circuit et des services multimedias en mode paquet - Google Patents

Convergence entre des communications vocales en mode circuit et des services multimedias en mode paquet

Info

Publication number
EP1636977A2
EP1636977A2 EP04743779A EP04743779A EP1636977A2 EP 1636977 A2 EP1636977 A2 EP 1636977A2 EP 04743779 A EP04743779 A EP 04743779A EP 04743779 A EP04743779 A EP 04743779A EP 1636977 A2 EP1636977 A2 EP 1636977A2
Authority
EP
European Patent Office
Prior art keywords
call
message
terminal
media
gateway
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.)
Withdrawn
Application number
EP04743779A
Other languages
German (de)
English (en)
Inventor
Bany Sylvain
Raheel Yuhanna
Alberto R. Villarica
Gregory T. Osterhout
Craik R. Pyke
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Rockstar Consortium US LP
Original Assignee
Nortel Networks Ltd
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
Priority claimed from US10/746,432 external-priority patent/US7899164B2/en
Priority claimed from US10/746,419 external-priority patent/US7894581B2/en
Application filed by Nortel Networks Ltd filed Critical Nortel Networks Ltd
Priority to EP10181944A priority Critical patent/EP2323359A1/fr
Publication of EP1636977A2 publication Critical patent/EP1636977A2/fr
Withdrawn legal-status Critical Current

Links

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
    • H04M7/0027—Collaboration services where a computer is used for data transfer and the telephone is used for telephonic communication
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04L—TRANSMISSION OF DIGITAL INFORMATION, e.g. TELEGRAPHIC COMMUNICATION
    • H04L12/00—Data switching networks
    • H04L12/64—Hybrid switching systems
    • H04L12/6418—Hybrid transport

Definitions

  • the present invention relates to communications, and in particular to associating traditional circuit-switched voice calls with multimedia services provided over a packet network.
  • packet-based voice sessions generally do not provide the level of quality or reliability as that provided by the circuit-switched networks.
  • the end users of a multimedia session will generally independently set up a voice call to correspond to their multimedia sessions, wherein there is no association between the multimedia sessions and the voice call.
  • the present invention provides a service node to assist in routing circuit-switched or packet-based calls to support voice communications.
  • the service node will recognize an attempt to initiate a call from a first terminal to a second terminal, and automatically provide information to media clients associated with the first and second terminals such that a media session can be readily established between the media- clients in association with the call.
  • the media session may support any type of service.
  • the service node may be configured to interact with telephony switches that support the first or second terminals, directly or indirectly via a signaling adaptor.
  • the signaling adaptor will provide the necessary message conversion from a first protocol used to communicate with the telephony switch to a second protocol used to communicate with the service node.
  • the service node will recognize an attempt to initiate a call and will route the call to a gateway, which is controllable by the service node. Once the call is sent to the gateway, the service node may provide instructions to the gateway for routing or otherwise processing the call. In either embodiment, the service node may include call routing logic, which is defined by a user of one of the terminals or media clients to control how the call is processed.
  • FIGURE 1 is a block representation of a communication environment according to one embodiment of the present invention.
  • FIGURES 2A-2C provide a communication flow for establishing a voice call over a circuit-switched network and a multimedia session over a packet network in association with one another according to one embodiment of the present invention.
  • FIGURE 3 is a block representation of a communication environment according to a second embodiment of the present invention.
  • FIGURES 4A-4C provide a communication flow for establishing a voice call over a circuit-switched network and a multimedia session over a packet network from a single multimedia client according to one embodiment of the present invention.
  • FIGURE 5 is a communication environment according to a third embodiment of the present invention.
  • FIGURES 6A-6C provide a communication flow illustrating an exemplary call routing process according to one embodiment of the present invention.
  • FIGURE 7 is a block representation of a service node according to one embodiment of the present invention.
  • FIGURE 8 is a block representation of a signaling adaptor according to one embodiment of the present invention.
  • FIGURE 9 is a block representation of a gateway according to one embodiment of the present invention.
  • the present invention facilitates control and association of voice sessions and multimedia sessions for multimedia services.
  • a communication environment 10 in which voice calls and multimedia sessions may be associated is illustrated according to a first embodiment.
  • end users will have corresponding media clients 12 (A and B), which may take the form of a personal computer, personal digital assistant, or like computing device, and may be configured to establish a multimedia session (A) with each other over a centralized packet network 14 and the corresponding access networks 16 (A and B).
  • the end users will also be associated with telephony terminals 18 (A and B), which are capable of establishing voice calls (B) therebetween over the Public Switched Telephone Network (PSTN) 20 via the corresponding telephony switches 22 (A and B).
  • PSTN Public Switched Telephone Network
  • the telephony switches 22 may support wired or wireless communications in a circuit-switched or packet- based fashion.
  • a signaling network 24 is used to provide call signaling to the telephony switches 22 to establish and tear down circuit-switched connections between the telephony switches 22 to support the voice call over the PSTN 20.
  • the signaling network 24 may take the form of a Signaling Systems 7 (SS7) network.
  • the telephony switches 22 may also be associated with corresponding gateways 26 (A and B), which may facilitate a portion of a voice call over the packet network 14.
  • the gateways 26 will effectively facilitate interworking between the PSTN 20 or telephony switches 22 and the packet network 14, wherein circuit-switched connections are transformed into packet sessions, and vice versa.
  • the gateway 26 will have a packet interface for communicating with the packet network 14, and a telephony interface, such as a Primary Rate Interface (PRI) for facilitating a circuit-switched connection with the telephony switch 22. Further detail regarding the operation of the gateways 26 (A and B) will be provided in association with other embodiments, which are described later in this specification.
  • a service node 28 is provided to communicate with the media clients 12 to assist in the establishment and control of the media session, as well as provide call signaling to assist in the control of the voice call.
  • the service node 28 may communicate with the media clients 12 through the packet network 14 and the associated access network 16 using the Session Initiation Protocol (SIP) or like media session control protocol.
  • the service node 28 may communicate with call control entities in the signaling network 24, the telephony switches 22, and the gateways 26, directly or indirectly, using SIP or like session control protocol.
  • the service node 28 can use SIP to communicate with the gateways 26 and the multimedia clients 12 in a direct manner, whereas a signaling adaptor 30 is used to convert SIP messages to Intelligent Network (IN) protocol messages to control call control entities in the signaling network 24 or the telephony switches 22.
  • IN Intelligent Network
  • Telephony switch 22A will receive the DTMF digits corresponding to directory number DN2 (step 100) and recognize that calls originating from telephony terminal 18A require the call control assistance of the service node 28. As such, telephony switch 22A may be provisioned to send an IN Offhook Delay message toward the service node 28.
  • the IN Offhook Delay trigger will be received by the signaling adaptor 30 (step 102), which will convert it into a corresponding SIP Notify message, which is sent to the service node 28 (step 104).
  • the SIP Notify message will identify the event that triggered the message and the directory number (DN2) for the called party.
  • the service node 28 may send a SIP Notify message to media client 12A indicating that the call log should be updated to include a call to directory number DN2 (step 106).
  • Media client 12A will update the call log to include the call to directory number DN2 as the latest outgoing call (step 108) and respond to the service node 28 with a SIP 200 OK message (step 110).
  • the service node 28 may also control the routing of the call, and will thus instruct telephony switch 22A how to proceed with establishing and routing the call initiated from telephony terminal 18A.
  • Telephony switch 22B may be provisioned to recognize that the service node 28 may control calls directed to telephony terminal 18B. As such, telephony switch 22B will send an IN Termination Attempt Trigger (TAT) toward the service node 28 to indicate a call is being routed to telephony terminal 18B.
  • TAT IN Termination Attempt Trigger
  • the signaling adaptor 30 will receive the IN TAT, which identifies the directory numbers for the calling and called parties (step 118), and will send a SIP Invite message to the service node 28 indicating a call is being initiated from directory number DN1 to directory number DN2 (step 120).
  • the service node 28 will execute any service node logic used to control the routing of an incoming call to telephony terminal 18B to decide how to respond to the SIP Invite message (step 122). In this example, assume the service node 28 decides to allow the call to continue toward telephony terminal 18B, and as such, will send a SIP 302 Moved Temporarily message back toward telephony switch 22B.
  • the SIP 302 Moved Temporarily message effectively identifies the next step for telephony switch 22B to take in routing the call.
  • the next step is to continue routing the call toward directory number DN2.
  • the signaling adaptor 30 will receive the SIP 302 Moved Temporarily message (step 124) and send an IN Authorize_Termination message to telephony switch 22B (step 126).
  • the IN Authorize_Termination message instructs telephony switch 22B to establish a 5 connection with telephony terminal 18B.
  • Telephony switch 22B will then initiate ringing of telephony terminal 18B (step 128), as well as send an ISUP Address Complete Message (ACM) to telephony switch 22A (step 130) to indicate telephony terminal 18B is ringing.
  • ACM ISUP Address Complete Message
  • the service node 28 can take the necessary steps 0 to arm the associated media clients 12 with information sufficient to establish a media session between them. Accordingly, the service node 28 may send a SIP Invite message identifying the address of multimedia client 12A (Client A) to media client 12B, which has an address of Client B (step 132). The SIP Invite message will alert media client 12B that a voice call is being initiated 5 from telephony terminal 18A toward its associated telephony terminal 18B.
  • Media client 12B will respond by sending a SIP 200 OK message to the service node 28 (step 134), as well as display a message to the user indicating that a call is coming in from directory number DN1 and providing any other associated call information (step 136).
  • the service node 28 will send a SIP Invite message to media client 12A using address Client A to identify the address (Client B) of media client 12B, as well as providing the • directory number (DN2) for telephony terminal 18B (step 138).
  • Media client 12A will respond by sending a SIP 200 OK to the service node 28 (step 140), as well as displaying any relevant call information to the user of media client 12A (step 142).
  • the call information may identify the called party as well as indicate that the call is in progress.
  • telephony switch 22B will receive an Offhook signal (step 144) and send an ISUP Answer Message (ANM) toward telephony switch 22A (step 146).
  • ANM ISUP Answer Message
  • a voice connection is established between telephony terminals 18A and 18B through telephony switches 22A and 22B (step 148).
  • a voice call is established between telephony terminals 18A and 18B, and media clients 12A and 12B are armed with sufficient information to initiate a media session therebetween.
  • the media session could be set up automatically or initiated by the user. Assume that the user of media client 12B decides to initiate an application sharing session with the user of media client 12A.
  • the SIP 200 OK message is received by the service node 28 (step 154), which will send a like SIP 200 OK message to media client 12B (step 156).
  • the media clients 12A and 12B can establish an application sharing session (step 158).
  • the establishment of a voice call results in automatically configuring corresponding media clients to support a media session. Once a particular media service is selected, a corresponding media session may be readily established.
  • the service node 28 will function to cancel the session and update presence information to indicate that the telephony terminals 18 are idle. Assume that telephony terminal 18B goes on hook, and telephony switch 22B determines that telephony terminal 18B has gone on hook (step 160). In response, telephony switch 22B will send a Termination Notification toward the service node 28. The Termination Notification is received by the signaling adaptor 30 (step 162), which will send a corresponding SIP Notify message to the service node 28 indicating that the call to directory number DN2 has been released (step 164).
  • the service node 28 may determine that telephony terminal 18B is no longer in use and send a SIP Notify message to media client 12B to indicate that telephony terminal 18B is currently idle, and that the call has been released (step 166). Media client 12B will log this information and respond with a SIP 200 OK message (step 168). In the meantime, telephony switch 22B will send an ISUP Release (REL) message toward telephony switch 22A (step 170). [0027] Similarly, telephony switch 22A will send a Termination Notification message toward the service node 28.
  • REL ISUP Release
  • the signaling adaptor 30 will receive the Termination Notification (step 172) and send a corresponding SIP Notify message toward the service node 28 indicating that the call from directory number DN 1 has been released (step 174).
  • the service node 28 will determine the presence information for telephony terminal 18A as being idle, and send a SIP Notify message indicating that the call involving telephony terminal 18A (directory number DN1) has been released, and that telephony terminal 18A is idle (step 176).
  • Media client 12A will log this information and respond to the service node 28 with a SIP 200 OK message (step 178).
  • the service node 28 may play a pivotal role in establishing and ending a media session in association with a voice call, provide call log information to an associated media client 12, and track and provide presence information bearing on the state of a user's telephony devices 18 or her relative availability for communications.
  • media client 12A is capable of supporting various types of media sessions, including a voice session (C) that may be facilitated in part over the PSTN 20 and terminate at telephony terminal 18B via telephony switch 22B.
  • Media client 12A will include a user interface to facilitate bi-directional voice communications, and as such will include a microphone and speaker and the necessary application software and hardware to support voice over packet (VoP) communications.
  • voice session (C) established between media client 12A and telephony terminal 18B
  • other media sessions (D) may be established in association with the voice session between media client 12A and media client 12B.
  • the voice session (C) includes a packet portion and a circuit-switched portion.
  • the packet portion is established between media client 12A and gateway 26A over access network 16A and the packet network 14, and the circuit-switched portion is established between gateway 26A and telephony terminal 18B via the PSTN 20 and telephony switch 22B.
  • the service node 28 will interact with gateway 26A and telephony switch 22B, via the signaling adaptor 30, to establish the voice session.
  • FIG. 4A-4C An exemplary communication flow for establishing the voice and media sessions between media client 12A and telephony terminal 18B, and media client 12A and media client 12B, respectively, is illustrated in Figures 4A-4C.
  • media client 12A decides to initiate a voice session between media client 12A and telephony terminal 18B.
  • media client 12A Upon receiving the appropriate instructions from the user, media client 12A will send a SIP Invite message indicating a voice session should be established from media client 12A (From Address: Client A) to directory number DN2.
  • the service node 28 is acting as a SIP proxy, the SIP Invite is received by the service node 28 (step 200), which will then send a like SIP Invite message to gateway 26A over the packet network 14 (step 202).
  • Gateway 26A will then take the necessary steps to instruct telephony switch 22B to establish a circuit-switched connection to telephony terminal 18B, which is associated with directory number DN2. If gateway 26A provides a Primary Rate Interface, a PRI Setup message may be sent directly or via the PSTN to telephony switch 22B to initiate the circuit-switched connection (step 204). If the circuit-switched connection needs to transit via the PSTN, the PRI messages may be converted to corresponding ISUP messages as is well known in the art.
  • Telephony switch 22B is again provisioned to alert the service node 28 of incoming calls to directory number DN2, and thus will send an IN TAT to the signaling adaptor 30 identifying the directory number DN2 for telephony terminal 18B and the connection information at the PRI of gateway 26A (step 206).
  • the signaling adaptor 30 will forward a corresponding SIP Invite to the service node 28 identifying the PRI connection information for gateway 26A and directory number DN2 for telephony terminal 18B (step 208).
  • the service node 28 will provide service node logic to determine how to route the call (step 210) and provide appropriate instruction to telephony switch 22B via the signaling adaptor 30.
  • the service node 28 may send a SIP 302 Moved Temporarily message instructing telephony switch 22B to terminate the call at directory number DN2 to the signaling adaptor 30 (step 212), which will send an IN Authorize Termination message to telephony switch 22B (step 214).
  • Telephony switch 22B will then initiate ringing of telephony terminal 18B (step 216), and send a PRI Ringing message back to gateway 26A to indicate that telephony terminal 18B is ringing (step 218).
  • gateway 26A will send a SIP 180 Trying message to the service node 28 to indicate that telephony terminal 18B is ringing (step 220).
  • the service node 28 will then take the necessary steps to prepare media clients 12A and 12B for a media session associated with the voice session.
  • the service node 28 may send a SIP Invite message to media client 12B using address Client B to identify the address Client A for media client 12A (step 222).
  • media client 12B Upon receipt, media client 12B will respond with a SIP 200 OK message (step 224).
  • Media client 12B may also display an alert to the user of media client 12B that an incoming call is being attempted at telephony terminal 18B and provide any call information associated therewith (step 226).
  • the call information may be sent in the SIP Invite message.
  • the service node 28 will also send a SIP Invite message to media client 12A using address Client A to provide the address Client B of media client 12B, as well as providing related call information to media client 12A (step 228).
  • Media client 12A will send a SIP 200 OK message in response to the SIP Invite (step 230), as well as providing an alert to the user of media client 12A (step 232).
  • the alert may provide call information pertaining to the voice session being established between media client 12A and telephony terminal 18B.
  • telephony switch 22B will receive an Offhook signal (step 234) and will send a PRI Connect message to gateway 26A (step 236).
  • media client 12B Upon being instructed to initiate the application sharing session, media client 12B will send a SIP Invite message toward media client 12A to initiate a media session to support application sharing.
  • the service node 28 may act as a SIP proxy, and receive the SIP Invite message on behalf of media client 12A (step 244) and forward a like SIP Invite message to media client 12A (step 246).
  • the SIP Invite message will include any address and port information for the respective media clients 12, as well as including an indication that the session to be established is an application sharing session.
  • SDP is again used to identify the session as an application sharing session.
  • media client 12A will send a SIP 200 OK message toward media client 12B.
  • telephony switch 22B will send a PRI Release message to gateway 26A (step 264), which will send a SIP Bye message to the service node 28 (step 266).
  • the service node 28 will then send a SlP Bye message to media client 12A (step 268).
  • Media client 12A will send a SIP 200 OK message back to the service node 28 (step 270), which will in turn send a SIP 200 OK message to gateway 26A (step 272), wherein the packet and circuit-switched portions of the voice session are ended.
  • the service node 28 will then send a SlP Bye message to media client 12A to indicate that the application sharing session between media clients 12A and 12B should end (step 274).
  • Media client 12A will respond with a SIP 200 OK message (step 276).
  • the service node 28 will also send a SlP Bye message to media client 12B indicating that the application sharing session between media clients 12A and 12B should end (step 278).
  • Media client 12B will then send a SIP 200 OK message back to the service node 28 (step 280).
  • the present invention may also facilitate the establishment and association of media sessions from one media client to multiple endpoints, wherein one endpoint may support a voice session and other endpoints may support other types of media sessions.
  • media sessions may be established prior to a voice session being established, under the control of the service node 28.
  • another embodiment of the present invention facilitates the transfer of call control from a traditional entity in the signaling network 24 to the service node 28, such that more advanced call processing functionality can be implemented.
  • the service node 28 may provide logic to control forwarding or rerouting of calls, in a virtually unlimited fashion in light of rules established by the telephony subscriber. Examples of such call routing may IB2004/002046
  • a significant portion of call routing control is transferred to the service node 28, wherein the service node 28 cooperates with gateway 26B to selectively route an incoming call to a desired destination according to a predefined set of rules implemented by the service node 28.
  • a call is initiated from telephony terminal 18A to telephony terminal 18B.
  • control of the call is transferred to the service node 28 by routing the call through gateway 26B (E).
  • the service node 28 From gateway 26B, the service node 28 will initially attempt to terminate the call at telephony terminal 18B (F), and if the call is not answered within a certain number of rings, the service node 28 will have the call forwarded to a voicemail system 32 (G).
  • G voicemail system 32
  • FIG. 6A-6C a communication flow is provided wherein call routing control is transferred to the service node 28, and the incoming call is initially routed to telephony terminal 18B for a select number of rings, and if unanswered, is forwarded to the voicemail system 32.
  • telephony switch 22A will receive the DTMF digits corresponding to directory number DN2 from telephony terminal 18A to indicate a call is being initiated to telephony terminal 18B (step 300).
  • Telephony switch 22A will send an ISUP IAM to telephony switch 22B indicating that a call is being initiated from directory number DN 1 to directory number DN2 (step 302).
  • Telephony switch 22B will recognize that the service node 28 should handle call processing for calls intended for directory number DN2, and will send an IN TAT toward the service node 28 via the signaling adaptor 30.
  • the IN TAT will identify the directory numbers for telephony terminals 18A and 18B.
  • the signaling adaptor 30 will receive the IN TAT (step 304) and send a SIP Invite message to the service node 28 indicating a voice call is being attempted between directory numbers DN1 and DN2 (step 306).
  • the service node 28 will provide service node logic to process the call (step 308) and will recognize that a complex call routing ruleset is in place for telephony terminal 18B.
  • Temporarily message including the call service directory number to the signaling adaptor 30 (step 310), which will send an IN Forward Call message to telephony switch 22B instructing telephony switch 22B to forward the incoming call to the call service directory number (step 312).
  • the service node 28 may be configured to send a SIP Invite message to media client 12B to provide information indicating that an incoming call from telephony terminal 18A is being attempted to telephony terminal 18B (step 314).
  • Media client 12B may respond with a SIP 200 OK message (step 316), as well as displaying any call information associated with the incoming call to the user of media client 12B (step 318).
  • the SIP Invite message will also identify in a History field the directory number DN2 for telephony terminal 18B.
  • the service node 28 will respond by sending a SIP 180 Trying message to gateway 26B (step 324), which will send a PRI Ringing message to telephony switch 22B (step 326) which will send an ISUP ACM to telephony switch 22A (step 328).
  • the PRI Setup message will also include the directory number DN 1 of the originating telephony terminal 18A and the OCN information indicating that the call was originally intended for directory number DN2.
  • Telephony switch 22B will recognize an incoming call intended for directory number DN2, and will again check with the service node 28 for routing instructions. Accordingly, an IN TAT is sent to the signaling adaptor 30 identifying the directory numbers for telephony terminal 18A and 18B 1 as well as the OCN information (step 338).
  • the signaling adaptor 30 will send a SIP Invite message to the service node 28 indicating that a call is being attempted from directory number DN1 to directory number DN2, and that the call was originally intended for directory number DN2 (step 340).
  • the service node 28 will again provide service node logic to process the call (step 342) and will determine that the call should be routed to directory number DN2. As such, the service node 28 will send a SIP 302 Moved Temporarily message, identifying directory number DN2 as the directory number to which the call should be routed, to the signaling adaptor 30 (step 344), which will forward an IN Continue message to telephony switch 22B (step 346). Telephony switch 22B will send a PRI Ringing message to Port 2 of gateway 26B (step 348), and initiate ringing of telephony terminal 18B (step 350).
  • the service node logic dictates that if the call to telephony terminal 18B is not answered within N rings, the call should be routed to the voicemail system 32. Accordingly, the service node logic may initiate a timer, which will expire in a time period corresponding to the N number of rings if it does not receive indication that the call has been answered. Assume that the call is not answered, and that the timer initiated by the service node logic expires (step 352). The service node 28 will then send a SIP Bye message to Port 2 of gateway 26B (step 354), which will send a PRI Release message to telephony switch 22B (step 356) to end the attempt to terminate the call at telephony terminal 18B.
  • Gateway 26B will respond with a SIP 180 Trying message (step 360), and send a PRI Setup message from Port 2 to telephony switch 22B (step 362).
  • the PRI Setup message will identify the voicemail directory number VM#, the originating directory number DN1 , and the OCN information identifying directory number DN2 as the originally called number.
  • Telephony switch 22B will then send a PRI Setup message to the voicemail system directory number VM# (step 364). Again, the PRI Setup message will identify the originating directory number DN 1 and the originally called number DN2.
  • the voicemail system 32 will receive the PRI Setup message and respond with a PRI Connect message, which is sent back to telephony switch 22B (step 366).
  • Telephony switch 22B will send a PRI Connect message to Port 2 of gateway 26B (step 368), which will send a SIP 200 OK message back to the service node 28 (step 370).
  • the SIP 200 OK message is in response to the SIP Invite message sent in step 358.
  • the service node 28 will then send a SIP 200 OK message to Port 1 of gateway 26B (step 372), which will send a PRI Connect message to telephony switch 22B (step 374).
  • Telephony switch 22B will then send an ISUP ANM to telephony switch 22A (step 376), wherein a voice connection is established between telephony terminal 18A and the voicemail system 32 through telephony switch 22A, telephony switch 22B, and ports 1 and 2 of gateway 26B (step 378).
  • the voicemail message has been left in the voicemail system 32 in association with a mailbox for the user of telephony terminal 18B, the user will hang up telephony terminal 18A, which will result in telephony switch 22A recognizing that telephony terminal 18A has gone on hook (step 380).
  • Telephony switch 22A will send an ISUP Release message to telephony switch 22B (step 382), which will send a PRI Release message to Port 1 of gateway 26B (step 384).
  • gateway 26B will send a SIP Bye message to the service node 28 (step 386), which will send a SIP Bye message to Port 2 of gateway 26B (step 388).
  • Port 2 of gateway 26B will send a PRI Release message back to telephony switch 22B (step 390), which will send a PRI Release message to the voicemail system 32 (step 392).
  • the third embodiment of the present invention allows the service node 28 to effectively transfer a call to a gateway 26B to facilitate advanced call processing, which may include implementing rules to control call forwarding, call routing, and the like.
  • the service node 28 can directly interact with the gateway 26B to implement the call routing and control logic for a given user.
  • the service node 28 may include a control system 34 having memory 36 with software 38 sufficient to provide the functionality described above.
  • the software 38 will include service node logic 40 capable of supporting the association of voice and media sessions, call routing, or a combination thereof.
  • the control system 34 will be associated with one or more communication interfaces 42 to facilitate communications with the media clients 12, gateways 26, and telephony switches 22, directly or indirectly via the signaling adaptor 30.
  • the service node functionality can be implemented in a standalone device or integrated with other entities on the packet network 14 or PSTN 20, such as within the telephony switches 22.
  • the signaling adaptor 30 may include a control system 44 with memory 46 having sufficient software 48 to implement the functionality described above.
  • the control system 44 will also be associated with one or more communication interfaces 50 to facilitate communications with the service node 28 and the telephony switches 22 or other entities in the signaling network 24.

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Abstract

Dans un mode de réalisation, un noeud de service va reconnaître une tentative visant à établir une communication entre un premier terminal et un deuxième terminal, et va fournir automatiquement des informations à des clients multimédias associés au premier et au deuxième terminal, de sorte qu'une session multimédia peut aisément être établie entre ces clients multimédias associés à la communication. Le noeud de service peut être configuré de manière à entrer en interaction avec les commutateurs téléphoniques gérant les premier et deuxième terminaux, directement ou indirectement par l'intermédiaire d'un adaptateur de signalisation. Dans un deuxième mode de réalisation, le noeud de service va reconnaître une tentative visant à établir une communication et va acheminer cette dernière à un noeud de transit gérable par le noeud de service. Une fois que la communication est envoyée au noeud de transit, le noeud de service peut donner des instructions à ce dernier pour qu'il achemine la communication ou la traite selon d'autres modalités.
EP04743779A 2003-06-19 2004-06-18 Convergence entre des communications vocales en mode circuit et des services multimedias en mode paquet Withdrawn EP1636977A2 (fr)

Priority Applications (1)

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EP10181944A EP2323359A1 (fr) 2003-06-19 2004-06-18 Convergence entre des communications vocales en mode circuit et des services multimédias en mode paquet

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US47971503P 2003-06-19 2003-06-19
US10/746,432 US7899164B2 (en) 2003-06-19 2003-12-24 Convergence of circuit-switched voice and packet-based media services
US10/746,419 US7894581B2 (en) 2003-06-19 2003-12-24 Convergence of circuit-switched voice and packet-based media services
PCT/IB2004/002046 WO2004112369A2 (fr) 2003-06-19 2004-06-18 Convergence entre des communications vocales en mode circuit et des services multimedias en mode paquet

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EP04743779A Withdrawn EP1636977A2 (fr) 2003-06-19 2004-06-18 Convergence entre des communications vocales en mode circuit et des services multimedias en mode paquet

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Also Published As

Publication number Publication date
WO2004112369A8 (fr) 2005-03-10
EP2323359A1 (fr) 2011-05-18
CA2768069C (fr) 2013-12-31
WO2004112369A2 (fr) 2004-12-23
WO2004112369A3 (fr) 2005-05-12
CA2529897A1 (fr) 2004-12-23
CA2529897C (fr) 2013-01-08
CA2768069A1 (fr) 2004-12-23

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