Source: http://www.freepatentsonline.com/9042372.html
Timestamp: 2019-04-19 14:38:35
Document Index: 637766344

Matched Legal Cases: ['Application No. 03', 'Application No. 03', 'Application No. 03817091', 'Application No. 03817091', 'Application No. 03817091', 'Application No. 03817091', 'Application No. 03817091', 'Application No. 99811317', 'Application No. 03817091', 'Application No. 02756674', 'Application No. 02756674', 'Application No. 05011176', 'Application No. 99946740', 'Application No. 02756674', 'Application No. 03734479', 'Application No. 05011176', 'Application No. 99946740', 'Application No. 3280', 'Application No. 2004', 'Application No. 2000', 'application No. 60']

Call transfer using session initiation protocol (SIP) - GENESYS TELECOMMUNICATIONS LABORATORIES, INC.
United States Patent 9042372
Zhakov, Vyacheslav I. (El Sobrante, CA, US)
Sayko, Vyacheslav V. (San Bruno, CA, US)
Tikin, Aleksandr V. (Walnut Creek, CA, US)
11/929766
H04M3/00; H04L12/56; H04L12/66; H04L29/06; H04L29/08; H04M3/51; H04M3/523; H04M3/58; H04M5/00; H04Q3/58; H04M3/42; H04M3/493; H04M3/56; H04M7/00; H04Q3/72
379/142.07, 379/212.01, 379/265.02, 370/352, 370/354, 370/355, 370/401
Download PDF 9042372 PDF help
7701925 Presence registration and routing node 2010-04-20 Mason et al. 370/352
7610384 Network telephony appliance and system for inter/intranet telephony 2009-10-27 Schulzrinne et al.
20080062974 Integrating SIP Control Messaging into Existing Communication Center Routing Infrastructure 2008-03-13 Kikinis et al.
20080049735 Methods and Apparatus for Accomplishing Call-State Synchronization and Event Notification between Multiple Private Branch Exchanges Involved in a Multiparty Call 2008-02-28 Kikinis et al.
7243162 Processing network communication control messages 2007-07-10 O'Neill et al.
20070121601 Integrating SIP Control Messaging into Existing Communication Center Routing Infrastructure 2007-05-31 Kikinis et al.
7133518 Methods and apparatus for accomplishing call-state synchronization and event notification between multiple private branch exchanges involved in a multiparty call 2006-11-07 Zhakov et al.
7120141 Integrating SIP control messaging into existing communication center routing infrastructure 2006-10-10 Kikinis
20060067309 Call transfer using session initiation protocol (SIP) 2006-03-30 Zhakov et al.
6958994 Call transfer using session initiation protocol (SIP) 2005-10-25 Zhakov et al.
6775269 Method and system for routing telephone calls between a public switched telephone network and an internet protocol network 2004-08-10 Kaczmarczyk et al.
6771639 Providing announcement information in requests to establish interactive call sessions 2004-08-03 Holden
6678735 Method and apparatus for a sip client manager 2004-01-13 Orton et al.
20030063590 Multimedia personalized call management (MPCM) 2003-04-03 Mohan et al.
20030021264 Call transfer using session initiation protocol (SIP) 2003-01-30 Zhakov et al.
20030002476 Integrated internet phone call routing system 2003-01-02 Chung et al. 370/352
20020150226 Caller treatment in a SIP network 2002-10-17 Gallant et al. 379/210.02
20020138603 Systems and methods for updating IP communication service attributes 2002-09-26 Robohm 709/223
20020126701 System and methods for using an application layer control protocol transporting spatial location information pertaining to devices connected to wired and wireless internet protocol networks 2002-09-12 Requena
6438114 Method and apparatus for enabling multimedia calls using session initiation protocol 2002-08-20 Womack et al.
20020110113 Switch with emulation client 2002-08-15 Wengrovitz 370/352
6389007 Method and apparatus for providing integrated routing for PSTN and IPNT calls in a call center 2002-05-14 Shenkman et al.
6363424 Reuse of services between different domains using state machine mapping techniques 2002-03-26 Douglas et al.
20020024943 Internet protocol based wireless call processing 2002-02-28 Karaul et al.
20020018464 Integrating SIP control messaging into existing communication center routing infrastructure 2002-02-14 Kikinis
6275574 Dial plan mapper 2001-08-14 Oran 379/201.01
6201804 Network telephony interface systems between data network telephony and plain old telephone service including CTI enhancement 2001-03-13 Kikinis
6137862 Failover mechanism for computer/telephony integration monitoring server 2000-10-24 Atkinson et al.
6055308 Method and system for determining and using multiple object states in a computer telephony integration system 2000-04-25 Miloslavsky et al.
6044146 Method and apparatus for call distribution and override with priority 2000-03-28 Gisby et al.
5974135 Teleservices computer system, method, and manager application for integrated presentation of concurrent interactions with multiple terminal emulation sessions 1999-10-26 Breneman et al.
5970126 Communication method and system 1999-10-19 Bowater et al.
5937051 Method and system for transferring calls and call-related data between a plurality of call centers 1999-08-10 Hurd et al. 379/212.01
5926539 Method and apparatus for determining agent availability based on level of uncompleted tasks 1999-07-20 Shtivelman
5848143 Communications system using a central controller to control at least one network and agent system 1998-12-08 Andrews et al.
5742675 Method and apparatus for automatically distributing calls to available logged-in call handling agents 1998-04-21 Kilander et al.
5546452 Communications system using a central controller to control at least one network and agent system 1996-08-13 Andrews et al.
5392345 Work at home ACD agent network 1995-02-21 Otto
AT426294T 2009-04-15 VERFAHREN UND VORRICHTUNG ZUR INTEGRIERTEN LEITWEGLENKUNG FUR PSTN UND IPNT ANRUFE IN EINEM ANRUFZENTRUM
AU752797B2 2002-10-03 Method and apparatus for providing integrated routing for PSTN and IPNT calls in a call center
AU2003238957A1 2003-12-31 Call transfer using session initiation protocol (sip)
CA2344832A1 2000-03-30 METHOD AND APPARATUS FOR PROVIDING INTEGRATED ROUTING FOR PSTN AND IPNT CALLS IN A CALL CENTER
CN1319298A 2001-10-24 System and method for managing power consumption in main station driving time-division duplex radio network
CN1205789C 2005-06-08 Integrated router, call center and call routing method
CN1669258B 2013-11-06 Call transfer using session initiation protocol (sip)
EP0866407 1998-09-23 System and method for telemarketing through a hypertext network
EP1421727 2004-05-26 INTEGRATING SIP CONTROL MESSAGING INTO EXISTING COMMUNICATION CENTER ROUTING INFRASTRUCTURE
EP1514374 2005-03-16 CALL TRANSFER USING SESSION INITIATION PROTOCOL (SIP)
EP1114541 2009-03-18 METHOD AND APPARATUS FOR PROVIDING INTEGRATED ROUTING FOR PSTN AND IPNT CALLS IN A CALL CENTER
EP2074772 2009-07-01 INTEGRATING SIP CONTROL MESSAGING INTO EXISTING COMMUNICATION CENTER ROUTING INFRASTRUCTURE
EP1619865 2010-02-03 Methods and apparatus for accomplishing call-state synchronization and event notification between multiple private branch exchanges
ES2321772T3 2009-06-10 METODO Y APARATO PARA PROPORCIONAR UN ENRUTADO INTEGRADO PARA LLAMADAS PSTN Y IPNT EN UN CENTRO DE LLAMADAS.
JP6477265 March, 1989
JP07170546 July, 1995
JP2000514985A 2000-11-07
JP2002525976A 2002-08-13
JP2005530394A 2005-10-06
WO1996027254A1 1996-09-06 COMMUNICATIONS SYSTEM AND METHOD FOR OPERATING SAME
WO1997018661A1 1997-05-22 INTELLIGENT INFORMATION ROUTING SYSTEM AND METHOD
WO1997050235A1 1997-12-31 TELECOMMUNICATIONS CALL CENTRE
WO/1998/031130 July, 1998 SYSTEM AND METHOD FOR OPERATING A PLURALITY OF CALL CENTERS
WO1998031130A1 1998-07-16 SYSTEM AND METHOD FOR OPERATING A PLURALITY OF CALL CENTERS
WO2000018074A1 2000-03-30 METHOD AND APPARATUS FOR PROVIDING INTEGRATED ROUTING FOR PSTN AND IPNT CALLS IN A CALL CENTER
WO2000072535A1 2000-11-30 ENTERPRISE CONTACT SERVER WITH ENHANCED ROUTING FEATURES
WO/2000/076158 December, 2000 NETWORK TELEPHONY APPLIANCE AND SYSTEM FOR INTER/INTRANET TELEPHONY
WO2000076158A1 2000-12-14 NETWORK TELEPHONY APPLIANCE AND SYSTEM FOR INTER/INTRANET TELEPHONY
WO2001035680A1 2001-05-17 METHOD FOR PROVIDING IP TELEPHONY WITH QS USING END-TO-END RSVP SIGNALING
WO2001043389A2 2001-06-14 DTMF DIGIT COLLECTION AND TRANSPORTATION FOR A PACKET NETWORK
WO2002039692A2 2002-05-16 PRESENCE WITH SPATIAL LOCATION INFORMATION
WO2003107575A2 2003-12-24 CALL TRANSFER USING SESSION INITIATION PROTOCOL (SIP)
WO2008045775A2 2008-04-17 INTEGRATING SIP CONTROL MESSAGING INTO EXISTING COMMUNICATION CENTER ROUTING INFRASTRUCTURE
JPH07170546A 1995-07-04
JPS6477265A 1989-03-23
J. Rosenberg et al., SIP for Presence, Memorandum, Internet Engineering Task Force, Nov. 13, 1998, 22 Pages, Bell Laboratories, Columbia University, XP-002325320.
M. Day et al., A Model for Presence and Instant Messaging, Memorandum, The Internet Society, Feb. 2000, 18 Pages, XP-002201444.
European Patent Office Action, Dated May 13, 2009, Issued in Patent Application No. 03 734 479.3-2414, 5 Pages.
European Patent Office Action, Dated Mar. 11, 2013, Issued in Patent Application No. 03 734 479.3-1858, 4 Pages.
Day, M. et al., “A Model for Presence and Instant Messaging,” Network Working Group, XP-002201444, Feb. 2000, 17 pgs.
Rosenberg, J., et al., “SIP for Presence,” Internet Engineering Task Force Internet Draft, XP-002325320, Nov. 13, 1998, 21 pgs.
International Search Report issued in corresponding International Application No. PCT/US03/18064, Jan. 12, 2004, 2 pgs.
Kikinis, Dan, Integrating SIP Messages Into Existing Call Centers, Disclosure Document #496199, dated Jun. 27, 2001, USPTO, 7 pages.
Chinese Office action with English Translation for Patent Application No. 03817091.4, dated Apr. 26, 2011, 13 pages.
Chinese Office action with English Translation for Patent Application No. 03817091.4, dated Aug. 3, 2012, 8 pages.
Chinese Office action with English Translation for Patent Application No. 03817091.4, dated Feb. 6, 2009, 9 pages.
Chinese Office action with English Translation for Patent Application No. 03817091.4, dated Jun. 26, 2009, 6 pages.
Chinese Office action with English Translation for Patent Application No. 03817091.4, dated Nov. 26, 2012, 7 pages.
Chinese Office action with English Translation for Patent Application No. 99811317.4, dated Jun. 18, 2004, 10 pages.
Chinese Rejection Decision with English Translation for Patent Application No. 03817091.4, dated Oct. 26, 2011, 9 pages.
European Office action and Search Report for 07843884.3, dated Apr. 21, 2010, 5 pages.
European Office action for Patent Application No. 02756674.4, dated Jan. 3, 2013, 7 pages.
European Office action for Patent Application No. 02756674.4, dated Mar. 5, 2010, 8 pages.
European Office action for Patent Application No. 05011176.4, dated Jun. 28, 2006, 4 pages.
European Office action for Patent Application No. 99946740.0, dated May 18, 2006, 5 pages.
European Search Report for Application No. 02756674.4,dated Oct. 30, 2009, 3 pages.
European Search Report for Patent Application No. 03734479.3, dated Nov. 21, 2005, 3 pages.
European Search Report for Patent Application No. 05011176.4, Nov. 25, 2005, 3 pages.
European Search Report for Patent Application No. 99946740.0, dated Nov. 25, 2004, 3 pages.
Handley, M. et al., RFC 2543-SIP: Session Initiation Protocol, Bell Labs, Mar. 1999, 153 pages.
Indian First Examination Report for Patent Application No. 3280/DELNP/2007, dated Feb. 27, 2014, 2 pages.
International Preliminary Examination Report for PCT/US02/23720, dated Apr. 2, 2004, 7 pages.
International Preliminary Examination Report for PCT/US99/20258, dated Nov. 14, 2000, 6 pages.
International Preliminary Report on Patentability for PCT/US2007/080535, dated Apr. 7, 2009, 5 pages.
International Search Report for PCT/US02/23720, mailed on Dec. 17, 2002, 1 page.
International Search Report for PCT/US07/80535, dated Apr. 17, 2008, 1 page.
International Search Report for PCT/US99/20258, mailed on Dec. 14, 1999, 2 pages.
Japanese Office action with English Translation for Patent Application No. 2004-514255, mailed on Nov. 21, 2006, 7 pages.
Japanese Office action with English Translation Patent Application No. 2000-571615, mailed Oct. 8, 2002, 5 pages.
Hoque, Nafiz E.
The present invention is a continuation of patent application Ser. No. 11/257,896, filed Oct. 24, 2005, which is a continuation of patent application Ser. No. 10/242,250, filed on Sep. 11, 2002 and issued as U.S. Pat. No. 6,958,994 on Oct. 25, 2005, which claims the benefit of Provisional Patent application No. 60/389,703, filed Jun. 17, 2002. A document disclosure in the DD program, number 496199 dated Jun. 19, 2001, was filed in case 09/927,301.
1. A system for automatic communication event routing and transfer capability in a multi-site communication-center environment comprising: a first processor in a first communication-center site, the first processor being configured to send a routing request to a second processor in a second communication-center site for transferring a communication event to the second communication-center site, wherein the routing request adheres to a proprietary protocol other than a standards-based protocol; a processing device coupled to the first and second processors, the processing device being configured to: receive a first message from the first processor, wherein the first message adheres to the standards-based protocol; and send a second message to the second processor for determining presence state of a target resource in the second communication-center site prior to transfer of the event, wherein the second message adheres to the standards-based protocol; and a switch coupled to the first processor for transferring the communication event to the second communication-center site, wherein the routing request and the first message are each transmitted by the first processor in response to receipt of the communication event at the first communication-center site, wherein the second processor at the second communication-center site is configured to transmit a response to the first processor indicative of whether the transfer of the communication event is to be completed, wherein the response is based on processing the routing request and the second message by the second processor.
2. The system of claim 1, wherein the standards-based protocol is a session initiation protocol (SIP).
3. The system of claim 2, wherein the second message that adheres to SIP includes parameters concerning requested call initiation procedures and permission for transferring the event.
4. The system of claim 2, wherein the standards-based protocol is an instant messaging and presence protocol (IMPP).
5. The system of claim 2, wherein the second message is configured to prompt a SIP based response indicative as to whether transfer is permitted.
6. The system of claim 5, wherein the routing request is configured to prompt a response indicative of whether transfer is permitted, wherein the response is based on the SIP based response.
7. The system of claim 1, wherein the second message and the routing request each includes an identifier associated with the event.
8. The system of claim 7, wherein the second processor is configured to match the identifier included in the second message and in the routing request for identifying the corresponding event.
9. The system of claim 1, wherein the second message is configured to be sent via a first data link and the routing request is configured to be sent via a second data link different from the first data link.
10. The system of claim 1, wherein the event handling by the processing device is configured to occur concurrently with the sending of the routing request by the first processor.
11. The system of claim 1, wherein the routing request is configured to invoke a routing rule for routing the event to an agent at the transfer destination.
12. The system of claim 1, wherein the routing request bypasses the processing device.
13. The system of claim 1, wherein the routing request is separate from the first message.
14. A method for automatic communication event routing and transfer capability in a multi-site communication-center environment comprising: sending a routing request by a first processor in a first communication-center site to a second processor in a second communication-center site for transferring a communication event to the second communication-center site, wherein the routing request adheres to a proprietary protocol other than a standards-based protocol; and invoking a processing device coupled to the first and second processors, wherein the processing device is configured to: receive a first message from the first processor, wherein the first message adheres to the standards-based protocol; and second a second message to the second processor for determining presence state of a target resource in the second communication-center site prior to transfer of the event, wherein the second message adheres to the standards-based protocol, wherein the routing request and the first message are each transmitted by the first processor in response to receipt of the communication event at the first communication-center site, wherein the second processor at the second communication-center site is configured to transmit a response to the first processor indicative of whether the transfer of the communication event is to be completed, wherein the response is based on processing the routing request and the second message by the second processor.
15. The method of claim 14, wherein the standards-based protocol is a session initiation protocol (SIP).
16. The method of claim 15, wherein the second message that adheres to SIP includes parameters concerning requested call initiation procedures and permission for transferring the event.
17. The method of claim 15, wherein the standards-based protocol is an instant messaging and presence protocol (IMPP).
18. The method of claim 15, wherein the second message prompts a SIP based response indicating whether transfer is permitted.
19. The method of claim 18, wherein the routing request is configured to prompt a response indicative of whether transfer is permitted, wherein the response is based on the SIP based response.
20. The method of claim 14, wherein the second message and the routing request each includes an identifier associated with the event.
21. The method of claim 20 further comprising: matching by the second processor the identifier included in the second message and in the routing request for identifying the corresponding event.
22. The method of claim 14, wherein the second message is sent via a first data link and the routing request is sent via a second data link different from the first data link.
23. The method of claim 14, wherein the routing request is sent by the first processor concurrently with the event handling by the processing device.
24. The method of claim 14, wherein the routing request invokes a routing rule for routing the event to an agent at the transfer destination.
The system above is used chiefly within a communication center and uses a proxy server to manage conversion between SIP protocol and T-server routing protocol. It has occurred to the inventor that many organizations that host communication centers host multiple center sites both based in telephony and data networks. It would be desirable to enable seamless communication event transfers including final destination routing between two or more sites that are network connected using unmodified standard protocols including SIP and instant message (IM)-based protocols.
One advantage with this example is that calls originating as IPNT calls within call center 17 may be sent as IPNT calls over data connection 25 or as converted COST calls over trunk 23. Another advantage is that LAN 55 is free to carry data other than IPNT audio packets.
FIG. 4 is an architectural overview of a communication network 401 wherein SIP messaging capability is integrated with routing infrastructure according to an embodiment of the present invention. Network 401 comprises a PSTN 414, a data—packet-network 417, which in this example is the well-known Internet network, and a telecommunications center 402.
In this example, there is a plurality of illustrated agent stations 610a-n within center 600 that are connected to LAN 601 for purpose of network communication. It will be appreciated that there will typically be, in actual practice, many more than the number of agent stations illustrated in this example. It may be assumed that each agent station 610a-n is at least adapted with a PC connected to LAN 601. In this embodiment an IP-capable telephone is available as well within each agent station 610a-n, the phones connected through, for example, a sound card interface to each PC.
In practice, communication events may arrive within center 600 at telephony switch 604 (COST) and at IR 609 (DNT). When an incoming event registers at the last switch within PSTN 608, TS software in processor 605 provides routing commands to the switch hosting the event before final routing. Processor 605 accesses SIP control SW 615 upon receiving notification over data network 615 of the presence of the pending event. The COST event is routed according to prevailing routing rules to central switch 604 within center 600. Data about the event arrives at processor 605 over network 615 and is passed on to the intended destination over link 613, through IR 609, over LAN 601 to the target one of agent stations 610a-n.
In this example of internal SIP-based routing, TS routine controls switch 604 in terms of routing protocol and destination. However event initiation, maintenance, and event termination from the point of processor 605 to any of the routing destinations is conducted according to SIP protocol. For example, assume that internal telephony wiring exists within center 600 and connects agent stations 610a-n (telephones) to switch 604. In this case TS routine would provide routing rules and commands while SIP protocol would be used to set up and notify one of a target agent's PC terminals that an incoming telephone call is being routed to the agent. The agent subscribes to TS to receive data about the caller or event and receives notification of the call and, perhaps a clickable link to the required data through SIP messaging and response interaction. In a preferred embodiment TS, SIP, and other protocols like IM type protocols can be extended between participating center sites.
FIG. 7 is an architectural overview of the communication center of FIG. 6 connected to an additional communication center 700 to illustrate SIP-based event transfer according to an embodiment of the invention. Incoming communication events arriving at either switch 604 or IR 609 may, in many cases, not be best handled within center 600 but may be better served by another cooperating communication center. Center 700, for purpose of discussion, is equipped identically as center 600 although this is not a requirement for successful practice of the present invention. For example, center 700 has a telephony switch 704 connected to a CTI processor 705, which in turn is connected to an IR 709. IR 709 is connected by a LAN connection 711 to a LAN 701, which supports agent stations 710a-n. A CIS server is not illustrated in this example, but may be assumed present in both centers 600 and 700.
Application server 702 may be hosted at the network level (PSTN or Internet), or may be hosted on a separate data network such as may be set up between cooperating instances of TS. In this case, server 702 is connected to processor 705 within center 700 by a data link 717, and to processor 605 within center 600 by a data link 718. In this embodiment, application server 702 may be accessed from a point on either LAN 601 or LAN 701 following the appropriate paths established for each center. For example, an agent operating one of stations 710a-n may access server 702 through link 711, IR 709, processor 705, and link 717. An agent operating one of stations 610a-n may access server 702 through link 611, IR 609, processor 605 and link 718.
In the case of an incoming event arriving at switch 604 in center 600, it may be determined that the best suited destination for the call is in fact an agent operating one of stations 710a-n in center 700. Caller ID can be established for the purpose of coordinating between TS request/response and SIP request/response. ANI services can be leveraged as well as many other identification and matching techniques to ensure that SIP and TS function for one event are appropriately associated with the event.
FIG. 8 is an architectural overview of the centers of FIG. 7 further enhanced for parlay through a communication server. In this example application server 702 now has a direct data connection to a communication server 801 maintained at the network level (Internet) or alternatively on a private network. This server may be, for example, a Microsoft Real-Time Communication (RTC) server. Communication server 801 is adapted in one embodiment to host multiparty sessions wherein connected parties may be operating a variety of communication devices and applications. The main function of communication server 801 is as a master server for initiating, establishing and tracking event-sessions using SIP. Communication server 801 accesses application server 702 for SIP functionality enabled by SIP application 713. Also connected to communication server 801 are processors 605 (center 600) and 705 (center 700). In this embodiment calling parties whether sourced from the PSTN or from the Internet network may be connected using SIP wherein server 801 maintains the active session control for all of the parties. Parties may terminate from the session or join the session as long as the session is established between at least two parties according to SIP rules.
At step 902a, the processor responsible for handling the event sends an SIP request to site 2, more particularly, the CTI processor (TS enhanced) enabling the communication center switch located within site 2. SIP generation within the processor of site 1 is enabled by an SIP control application running on a server accessible to both the processors of site 1 and cooperating site 2. The shared server is analogous to application server 702 of FIG. 7. The SIP request has at least a header and a body. The header contains caller identification information and event identification. The SIP body contains parameters concerning requested call initiation procedures and permission for transferring the event through, in this case, the PSTN network to the switch at site 2. Existing SIP and IMPP protocols are presence reporting capable. Therefore, presence information tailored to current availability states of agents or systems within a communication center can be entirely handled using SIP or IMPP messaging.
At step 902b the processor handling the event at site 1 sends a proprietary TS routing request to confirm routing protocol for the event and to enable passing of any additional pertinent data gleaned from the caller to the event handling processor at site 2. The process of step 902b occurs simultaneously with the process of step 902a. The routing request can be trimmed of requirements for requesting agent availability information because that can be handled through SIP or other presence protocols. If additional complex protocols exist for routing such as skill level identification, statistical routing information, historical or predictive routing information, then a TS request must be sent along with the SIP request.
Both the SIP request and the TS request are tagged with matching identification associated with the event pending transfer so that the event handling processor of site 2 can match the requests to the same event. Standard call identification procedures can be applied. In one embodiment random identification codes are generated at the time of request generation. Any additional data about the caller sent in the TS request at step 902b is also tagged with the appropriate identification.
It is noted herein that the network over which the SIP request is sent includes the shared application server in its path. The applications server coordinates and tracks the request/response interaction as well as session states for any session resulting from the event. By contrast, the TS request and data is sent over a separate network set up between the two active processors. This network may be a proprietary private network such as a virtual private network (VPN), or any other secure data network.
At step 903, the event handling processor at site 2 receives the requests sent in steps 902a and 902b. The requests are processed for appropriate responses. This may include consulting routing rules, and transferring hard data from TS to agent desktop on the TS side. Each response is generated in a cooperative manner so that there are no conflicts. For example, if the SIP response fails to permit transfer, the TS response must also indicate that transfer is not possible. In this way no conflicting determinations exist saving bandwidth. The event handling processor within site 2 sends the generated responses back to the event handling processor of site 1 giving a green light for the transfer.
<- Previous Patent (Integrating telephon...) | Next Patent (Data flow mobility) ->