Abstract:
A method and apparatus for call setup signaling in a VoIP network is disclosed. An application server receives call information from a network node, which causes the application server to be inserted in the call setup signaling path. The application server provides any special feature processing necessary for call setup. Upon the application server determining that it is no longer required in the signaling path for call setup, the application server removes itself from the call setup signaling path. The application server may remove itself from the signaling path prior to completion of call setup, thereby freeing up network resources and reducing post dial delay. In various embodiments, the application server removes itself from the call setup signaling path by transmitting information utilizing particular protocol messages.

Description:
This application claims the benefit of U.S. Provisional Application No. 60/465,374 filed Apr. 25, 2003, which is incorporated herein by reference. 
    
    
     BACKGROUND OF THE INVENTION 
     The conventional Public Switched Telephone Network (PSTN) is a circuit switched network in which calls are assigned dedicated circuits during the duration of the call. Such networks are well known in the art, and service providers have developed various services which may be provided to subscribers via such a conventional circuit switched network. 
     Recently, data packet networks, such as local area networks (LAN) and wide area networks (WAN) have become more prevalent. These data packet networks operate in accordance with the internet protocol (IP) and such networks are referred to as IP networks. The popularity of IP networks has created an interest in providing voice and related services over IP networks. 
     Conventional PSTN voice services dedicate a circuit connection between a calling and called party, and as such, that connection is guaranteed a certain level of performance because it is not shared with any other network users. IP networks, on the other hand, are shared networks in which the network resources are shared between users. The notion of a connection in a data packet network is very different from the notion of a connection in a circuit network. In a circuit network, the connection is a dedicated circuit which is used only by the calling and called parties. As such, it is easy to guarantee a certain level of service via the circuit network. The problem with such a network is that of efficiency. That is, the dedication of a circuit between all calling and called parties may be inefficient because such dedicated circuits provide more bandwidth than is necessary. In a data network, the connection between two parties is not dedicated, and traffic between the parties is transmitted via the data packet network along with the data packets of other users. There is no dedicated path between the parties, and data packets may be transmitted between the parties via different paths, depending upon network traffic. 
     In the PSTN, call setup is controlled by a signaling network in accordance with the well known Signaling System No. 7 (SS7). An SS7 network exists within the PSTN network and controls call setup by conveying labeled messages via signaling channels which are separate from the voice channels. The details of an SS7 network is well known and the details will not be described in further detail herein. 
     Signaling in a voice over IP (VoIP) network is accomplished by sending messages utilizing the Session Initiation Protocol (SIP) which will be described in further detail below. In contrast to the SS7 network utilized in the conventional PSTN network, SIP messages in a VoIP network are not transmitted via a dedicated signaling network, but are transmitted like any other data packets. 
     One of the goals of a voice network is to minimize post dial delay (PDD) which is the time required to connect the call after the user finishes dialing the called number. One of the factors that determines the PDD is the extent of signaling required to set up the call. Thus, a reduction in the required signaling will decrease the PDD and therefore increase customer satisfaction. Another goal of a voice network is to decrease the load on the various network elements. Once again, a reduction in the required signaling also reduces the load on the network signaling elements. 
     Therefore, what is needed is a method and apparatus for reducing the required signaling in a VoIP network. 
     BRIEF SUMMARY OF THE INVENTION 
     The present invention improves call setup signaling in an internet protocol network. In accordance with the invention, a network node removes itself from the call setup signaling path upon a determination that the network node is no longer needed in the signaling path in order to successfully complete the call setup. Since this removal may occur prior to completion of call setup, the removal of the network node frees up network resources, speeds up call setup, and reduces PDD. 
     In one embodiment, the network node is an application server which provides call feature processing during call setup. The application server receives call information during call setup and is thereby inserted into the signaling path. The application server then determines whether it is required in the signaling path in order to complete the call setup. If the application server is not required in the signaling path to complete the call setup, the application server removes itself from the signaling path. In various embodiments, the application server removes itself from the signaling path by transmitting particular signaling protocol messages as will be described in further detail below. If the application server is required in the signaling path to complete call setup, then the application server provides the required feature processing. Thereafter, the application server determines that it is no longer required in the signaling path and removes itself from the signaling path. Again, this removal may take place prior to completion of call setup, thereby freeing up network resources, speeding up call setup, and reducing PDD. 
     The principles of the present invention may be implemented in various call setup scenarios. For example, the principles of the invention are useful when the services of a media server are necessary for playing announcements and collecting user input during call setup. In such an embodiment, the application server which invokes the media server and validates the user input only remains in the call setup signaling path as long as necessary to validate the user input, and thereafter removes itself from the signaling path when it is no longer required in the signaling path to complete the call setup. 
     The principles of the present invention are also useful in an embodiment in which the application server is used to provide primary and alternate call routing numbers. In such an embodiment, the application server may remove itself from the signaling path after providing the alternate routing number. 
     As will be appreciated from the following detailed description, there are many call setup scenarios, in addition to those described herein, in which the principles of the present invention would be advantageous. 
     These and other advantages of the invention will be apparent to those of ordinary skill in the art by reference to the following detailed description and the accompanying drawings. 
    
    
     
       BRIEF DESCRIPTION OF THE DRAWINGS 
         FIG. 1  shows an IP network illustrating an embodiment of the present invention; 
         FIG. 2  is a flowchart showing the steps performed by an application server in accordance with one embodiment of the invention; 
         FIG. 3  shows an IP network illustrating an embodiment of the present invention; 
         FIG. 4  shows an IP network illustrating an embodiment of the present invention; and 
         FIG. 5  shows an IP network illustrating an embodiment of the present invention. 
     
    
    
     DETAILED DESCRIPTION 
       FIG. 1  shows an IP network in which one embodiment of the present invention may be implemented. The network utilizes the Session Initiation Protocol (SIP) in order to set up connections (e.g., VoIP calls) between users. SIP is a well known application-layer control protocol used to establish, modify and terminate sessions such as IP telephony calls. SIP is described in detail in Internet Engineering Task Force (IETF) Request for Comments (RFC) 3261; SIP: Session Initiation Protocol; J. Rosenberg, H. Schulzrinne, G. Camarillo, A. Johnston, J. Peterson, R. Sparks, M. Handley, E. Schooler; June 2002, which is incorporated by reference herein. The details of SIP will not be described herein, as the protocol is well known to those skilled in the art. The protocol will be described only insofar as necessary for an understanding of the present invention. 
     With reference to  FIG. 1 , it is to be understood that the network elements shown in  FIG. 1  are logical entities. Such logical entities may be implemented in various hardware configurations. For example, these network elements may be implemented using programmable computers which are well known in the art. Such programmable computers would have the required network interfaces to allow for network communication, as well as appropriate software for defining the functioning of the network elements. Such software is executed on one or more computer processors which control the overall operation of the network elements via execution of such software. The detailed hardware and software configuration of the network elements will not be described in detail herein. One skilled in the art of data networking and computers could readily implement such network elements given the description herein. As used herein, a network element refers to a logical entity which performs a network function. A network node refers to the computing platform on which a network element is implemented. 
     Referring now to  FIG. 1 , assume that IP enabled telephone  102  wishes to initiate an IP telephony call to IP enabled telephone  104 . In  FIG. 1 , telephone  102  is connected to a border element (BE)  106  which provides telephone  102  access to the IP network. Similarly, telephone  104  is connected to BE  108  which provides telephone  104  access to the IP network. In the example of  FIG. 1 , the transaction begins by telephone  102  sending an INVITE request  110  addressed to telephone  104 &#39;s Uniform Resource Identifier (URI) which identifies telephone  104 . The INVITE request contains a number of header fields which are named attributes that provide additional information about a message. The details of an INVITE are well known and will not be described in detail at this point. 
     The INVITE message  110  is received at the call control element (CCE)  112 . The CCE  112  performs the functions of interfacing with other network elements such as Border Elements (BE), Service Brokers (SB), Application Servers (AS), Media Servers (MS), Network Routing Engines (NRE), and others, to provide the necessary functions to process a call request. The CCE  112  determines whether special feature processing is required by the call by sending an INVITE message  130  to service broker (SB)  135 . Examples of special processing are 8YY (e.g., 800) calls or Software Defined Network (SDN) calls. The SB  135  determines whether special processing is required based on call information it receives in the INVITE message  130 . It is noted that while the SB function is a separate logical function from the CCE, the SB function may be contained in the CCE network element or a standalone network element. If special feature processing is required, the SB  135  determines the appropriate application server to provide the special feature processing for the call. The SB  135  sends a REDIRECT message  132  to the CCE  112  indicating the IP address of the appropriate application server to provide the feature processing. The CCE  112  sends a query (an SIP INVITE)  114  to the application server identified by the SB  135 , for example AS  116 . At this point, the AS  116  has been inserted into the signaling path for the call setup signaling being described herein. 
     The AS  116  contains the intelligence for offering intelligent network services such as local, toll-free, virtual private networks, and various multimedia features such as email and click-to-dial. In accordance with one embodiment of the invention, upon the AS  116  being inserted into the signaling path for the call setup, the AS  116  performs the steps shown in  FIG. 2 . First, the AS  116  determines in step  202  whether it is required in the signal path for call setup. At this point in the example, the AS  116  is required in the signaling path because the CCE  112  requires the routing number for the call. Therefore, control passes to step  204  in which the AS  116  executes service logic (e.g., computer program code) and performs the required feature processing. For the present example, assume that the only feature processing required is that the AS  116  provide a routing number for the call. The processing loop (i.e., steps  202  and  204 ) shown in  FIG. 2  represents that the AS  116  will continually determine whether it is still required in the signaling path. When the AS  116  determines that it is no longer required in the signaling path, then in step  206  the AS  116  removes itself from the signaling path. 
     In the example being described herein, after determining the routing number, the AS  116  will determine in step  202  that it is no longer required in the signaling path. Thus, in the particular embodiment being described, the AS  116  removes itself from the signaling path by returning the routing number to the CCE  112  by utilizing an SIP REDIRECT message  118 . The REDIRECT message  118  contains the routing number (as well as other call setup required information). As is well known, the SIP REDIRECT message returns the required information and removes the AS  116  from the signaling path for further call setup. In an alternate embodiment, the AS  116  removes itself from the signaling path by returning the routing number to the CCE  112  by utilizing an SIP REFER message. Again, like the REDIRECT message, the REFER message would contain the routing number (as well as other call setup required information) and would remove the AS  116  from the signaling path for further call setup. Briefly, and as is well known, a REDIRECT message is a 3xx response message generated by a redirect server user agent in response to received requests. The REDIRECT message directs the requesting client to contact an alternate set of URI&#39;s. A REFER message indicates that the recipient (identified by a Request-URI) should contact another network element using contact information provided in the Request-URI. 
     Upon receipt of the REDIRECT message, the CCE  112  sends another INVITE message  138  to the SB  135  to determine whether further feature processing is required. The SB  135  determines that no further special processing is required, and returns a REDIRECT message  140  to the CCE  112  directing the CCE  112  to the network routing engine (NRE)  122 . The CCE  112  sends an SIP INVITE  120  to the NRE  122  to determine the IP address of the appropriate BE for further routing. The NRE  122  determines the IP address of the appropriate BE using the routing information returned by the AS  116 . The NRE  122  returns the requested information by message  124 . It is noted that the NRE  122  is shown as a separate logical network element in the network of  FIG. 1 . In various embodiments, the NRE function may be implemented in the same network element as the CCE  112  or on a separate network element. 
     Upon receipt of the address of the appropriate BE (in this case BE  108 ), CCE  112  forwards INVITE message  126  to telephone  104  via BE  108 . The telephone  104  accepts the call by sending an OK message  128  back to the CCE  112 . The CCE  112  forwards the OK message  130  to telephone  102  via BE  106 . The VoIP call between telephone  102  and telephone  104  is now set up. 
     It is noted that in accordance with the above described embodiment of the invention, the AS  116  removes itself from the signaling path by returning REDIRECT message  118 . If the principles of the present invention were not used, then the message returned by the AS  116  could be an INVITE message instead of a REDIRECT message, in which case the AS  116  would remain in the call setup signaling path during the remainder of the call setup signaling operations. This would use additional network resources, and would increase PDD. By removing the application server from the signaling path in accordance with the principles of the invention, network resources are freed up and PDD is reduced. 
       FIG. 3  illustrates another embodiment of the present invention. Referring now to  FIG. 3 , in a manner similar to that described above in connection with  FIG. 1 , assume again that IP enabled telephone  302  wishes to initiate an IP telephony call to IP enabled telephone  304 . Telephone  302  is connected to a border element (BE)  306  which provides telephone  102  access to the IP network and telephone  304  is connected to BE  308  which provides telephone  304  access to the IP network. The transaction begins by telephone  302  sending an INVITE request  310  addressed to telephone  304 &#39;s Uniform Resource Identifier (URI). 
     The INVITE message  310  is received by CCE  312 . The CCE  312  determines whether special feature processing is required by the call by sending an INVITE message  330  to service broker (SB)  335 . In the present embodiment, assume that special feature processing is required to set up this call, and that the special feature processing required is that input is required from the user of telephone  302 . The SB  335  determines the appropriate application server to provide the special feature processing for the call. The SB sends a REDIRECT message  332  to the CCE  312  indicating the IP address of the appropriate application server to provide the feature processing. The CCE  312  sends a query (an SIP INVITE)  314  to the application server identified by the SB  335 , for example AS  316 . At this point, AS  316  has been inserted into the signaling path for the call setup signaling. Again, upon the AS  316  being inserted into the signaling path for the call setup, the AS  316  performs the steps shown in  FIG. 2 . In this example, the AS  316  is required in the signaling path because the call requires the collection of user input in order to be setup. Therefore, control passes to step  204  in which the AS  316  executes service logic and performs the required feature processing. 
     In accordance with this example, the AS  316  service logic indicates that the call setup requires the services of a media server (MS)  342 . The media server  342  provides the services of providing announcements and collecting information from a caller when features requiring caller interaction are required. AS  316  sends an INVITE  318  to the CCE  312 . The INVITE  318  contains a uniform resource locator (URL) of a script located at the MS  342 . The script identifies service logic necessary for call setup which is to be performed by MS  342 . Upon receipt of the INVITE  318 , the CCE  312  sends an INVITE  320  containing the information received in message  318  to the NRE  322 . The NRE  322  determines the IP address of the appropriate media server (i.e., MS  342 ). The NRE  322  returns the requested information by message  324 . 
     The CCE  312  then sends an INVITE message  344  to the MS  342 . The INVITE message  344  contains the URL of the script identified by the AS  316 . In one embodiment, the script could identify a Voice Extensible Markup Language (VoiceXML) script. VoiceXML is a protocol designed for creating audio dialogs that feature synthesized speech, digitized audio, recognition of spoken and DTMF key input, recording of spoken input, telephony, and mixed initiative conversations. VoiceXML is one type of script which may be executed by the MS  342 , and the details of such a script are not required for an understanding of the present invention. 
     Upon receipt of the INVITE message  344 , the MS  342  executes the script identified by the URL in the INVITE message  344 . The MS  342  may now play certain voice announcements directly to the telephone  304  via BE  308  using the early media protocol as represented by  346  which is a real time transport protocol (RTP) connection. Early media protocol is a protocol for exchanging media (e.g., audio) prior to a call being setup. Early media is well known, the details of which are not required for an understanding of the present invention. The MS  342  then plays an appropriate announcement to telephone  304 . In the example being discussed, it will be assumed that the announcement is a request for the user of telephone  304  to enter some information (e.g., an account number) using the telephone keypad. The MS  342  collects the entered information, and when the user input is complete, the MS  342  sends the collected information to the AS  316  via connection  348 , which is a hypertext transfer protocol (HTTP) connection as is well known in the art. 
     Upon receipt of the information from the MS  342 , the AS  316  validates the information, and if the information is validated, the AS  316  determines that call setup may proceed. Referring again to  FIG. 2 , the processing up to this point is represented by step  204 , during which the AS  316  has been providing feature processing. It is at this point in the processing (i.e., after collection and validation of user input) that AS  316  determines that it is no longer required in the signaling path (step  202 ) and in step  206  the AS  316  removes itself from the signaling path as follows. AS  316  cancels the INVITE message  318  using the SIP CANCEL message  350 . After canceling the INVITE, AS  316  then issues an SIP REDIRECT message  352  which redirects the initial INVITE message  314 . Thus, AS  316  is removed from the signaling path for further call setup. In an alternate embodiment, the AS  316  removes itself from the signaling path by utilizing an SIP REFER message. 
       FIG. 4  shows another signaling situation in which the principles of the present invention may be implemented. Referring now to  FIG. 4 , in a manner similar to that described above in connection with  FIG. 1 , assume again that IP enabled telephone  402  wishes to initiate an IP telephony call to IP enabled telephone  404 . Telephone  402  is connected to a border element (BE)  406  which provides telephone  402  access to the IP network and telephone  404  is connected to BE  408  which provides telephone  404  access to the IP network. The transaction begins by telephone  402  sending an INVITE request  410  addressed to telephone  404 &#39;s Uniform Resource Identifier (URI). 
     The INVITE message  410  is received by CCE  412 . The CCE  412  determines whether special feature processing is required by the call by sending an INVITE message  430  to SB  435 . In the present embodiment, assume that special processing is required to set up this call, and that the special feature processing required is that the call is eligible for special “overflow to an alternate number on egress busy” treatment. That is, if the initial egress point of the call is busy, the call may be transferred to another egress point. The SB  435  determines the appropriate application server to provide the special feature processing for the call. The SB sends a REDIRECT message  432  to the CCE  412  indicating the IP address of the appropriate application server to provide the feature processing. The CCE  412  sends a query (an SIP INVITE)  414  to the application server identified by the SB  435 , for example AS  416 . At this point, AS  416  has been inserted into the signaling path for the call setup. Again, upon the AS  416  being inserted into the signaling path for the call setup, the AS  416  performs the steps shown in  FIG. 2 . In this example, the AS  416  is required in the signaling path because the call may require the treatment in order to be setup. Therefore, control passes to step  204  in which the AS  416  executes service logic and performs the required feature processing. 
     In accordance with this example, the AS  416  service logic indicates that the call setup may require overflow treatment and therefore AS  416  must remain in the signaling path. Thus, AS  416  sends an INVITE  418  to the CCE  412 . The INVITE  418  contains the primary routing number for the call. Upon receipt of the INVITE  418 , the CCE  412  sends another INVITE message  438  to the SB  435  to determine whether further special processing is required. Now, the SB  435  will determine that no further special processing is required, and returns a REDIRECT message  440  to the CCE  412  directing the CCE  412  to the NRE  422 . The CCE  412  sends an SIP INVITE  420  to the NRE  422  to determine the IP address of the appropriate BE for further routing. The NRE  422  determines the IP address of the appropriate BE using the routing information returned by the AS  416 . The NRE  422  returns the requested information by message  424 . 
     Upon receipt of the address of the appropriate BE (in this case BE  408 ), CCE  412  forwards INVITE message  426  to telephone  404  via BE  408 . In accordance with the particular example of  FIG. 4 , assume that BE  408  returns a busy message  428  (e.g., “SIP  486  User Busy”) to the CCE  412 . This message  428  indicates that BE  408  cannot accept the call. The CCE  412  then forwards this busy message  428  to the AS  416  as message  442 . It is noted that the AS  416  is still in the call signaling path because it determined in step  202  ( FIG. 2 ) that it was still required in the signal path because overflow processing may be necessary. 
     Upon receipt of message  442 , AS  416  (in step  204 ), based on the service logic for this call, determines an alternate routing number associate with the call. After determining the alternate routing number, AS  416  determines in step  202  that it is no longer required in the signaling path because it has provided the overflow treatment processing. Since no further overflow processing may be performed with respect to this call, the AS  416  may remove itself from the signaling path in accordance with the principles of the present invention. Thus, AS  416  removes itself from the signaling path by returning the alternate number to the CCE  412  utilizing an “SIP  302  Moved Temporarily” message  444 . This SIP message removes AS  416  from the signaling path. 
     Upon receipt of the “SIP  302  Moved Temporarily” message  444 , the CCE  412  will send another INVITE message  446  to the SB  435  to determine whether further special feature processing is required. The SB  435  determines that no further special feature processing is required, and returns a REDIRECT message  447  to the CCE  412  directing the CCE  412  to the NRE  422 . The CCE  412  sends an SIP INVITE  448  to the NRE  422  to determine the IP address of the appropriate border element for further routing. The NRE  422  determines the IP address of the appropriate border element using the routing information returned by the AS  416  in message  444 . The NRE  422  returns the requested information by INVITE message  450 . Upon receipt of the address of the appropriate alternate BE (in this case BE  452 ), CCE  412  forwards the INVITE message  456  to telephone  454  via BE  452 . The telephone  454  accepts the call by sending an OK message  458  back to the CCE  412 . The CCE  412  forwards the OK message  460  to telephone  402  via BE  406 . The VoIP call between telephone  402  and alternate overflow telephone  454  is now set up. 
       FIG. 5  shows yet another signaling situation in which the principles of the present invention may be implemented. Referring now to  FIG. 5 , in a manner similar to that described above in connection with  FIG. 1 , assume again that IP enabled telephone  502  wishes to initiate an IP telephony call to IP enabled telephone  504 . Telephone  502  is connected to BE  506  which provides telephone  502  access to the IP network and telephone  504  is connected to BE  508  which provides telephone  504  access to the IP network. The transaction begins by telephone  502  sending an INVITE request  510  addressed to telephone  504 &#39;s Uniform Resource Identifier (URI). 
     The INVITE message  510  is received by CCE  512 . The CCE  512  determines whether special feature processing is required by the call by sending an INVITE message  530  to SB  535 . In the present embodiment, assume that special feature processing is required to set up this call, and again that the special feature processing required is that the call is eligible for special “overflow to an alternate number on egress busy” treatment. The SB  535  determines the appropriate application server to provide the special feature processing for the call. The SB  535  sends a REDIRECT message  532  to the CCE  512  indicating the IP address of the appropriate application server to provide the feature processing. The CCE  512  sends a query (an SIP INVITE)  514  to the application server identified by the SB  535 , for example AS  516 . At this point, AS  516  has been inserted into the signaling path for the call setup. Again, upon the AS  516  being inserted into the signaling path for the call setup, the AS  516  performs the steps shown in  FIG. 2 . In this example, the AS  516  is again required in the signaling path because the call may require overflow treatment in order to be setup. Therefore, control passes to step  204  in which the AS  516  executes service logic and performs the required feature processing. 
     In accordance with this example, the AS  516  service logic indicates that the call setup may require overflow treatment. Unlike the embodiment described above in connection with  FIG. 4 , here the AS  516  provides both the primary routing number as well as the alternate routing number for the call to the CCE  512 . Therefore, the AS  516  makes the determination (in step  202 ) that it is no longer required in the signaling path, and removes itself from the signaling path in step  206 . The AS  516  removes itself from the signaling path by providing the primary and alternate routing number to the CCE  512  utilizing an SIP REDIRECT message (e.g., SIP 3XX)  518 . 
     Upon receipt of the REDIRECT message  518 , the CCE  512  sends another INVITE message  538  to the SB  535  to determine whether further special processing is required. Now, the SB  535  determines that no further special processing is required, and returns a REDIRECT message  540  to the CCE  512  directing the CCE  512  to the NRE  522 . The CCE  512  sends an SIP INVITE  520  to the NRE  522  to determine the IP address of the appropriate border element for routing. The NRE  522  determines the IP address of the appropriate border element using the primary routing information returned by the AS  516 . The NRE  522  returns the requested information by message  524 . 
     Upon receipt of the address of the appropriate border element (in this case BE  508 ), CCE  512  forwards INVITE message  526  to telephone  504  via BE  508 . In accordance with the particular example of  FIG. 5 , assume that BE  508  returns a busy message  528  (e.g., “SIP  486  User Busy”) to the CCE  512 . This indicates that BE  508  cannot accept the call. The CCE  512  already has the alternate routing number and the CCE  512  sends another INVITE message  544  to the SB  535  to determine whether further special processing is required. The SB  535  will determine that no further special processing is required, and returns a REDIRECT message  546  to the CCE  512  directing the CCE  512  to the NRE  522 . The CCE  512  sends an SIP INVITE  548  to the NRE  522  to determine the IP address of the appropriate border element for further routing. The NRE  522  determines the IP address of the appropriate border element using the alternate routing information. The NRE  522  returns the requested information by INVITE message  550 . Upon receipt of the address of the appropriate alternate border element (in this case BE  552 ), CCE  512  forwards the INVITE message  556  to telephone  554  via BE  552 . The telephone  554  accepts the call by sending an OK message  558  back to the CCE  512 . The CCE  512  forwards the OK message  560  to telephone  502  via BE  506 . The VoIP call between telephone  502  and alternate overflow telephone  554  is now set up. 
     As may be seen from the above description, the present invention provides for an application server to remove itself from the signaling path upon a determination that it no longer needs to remain in the signaling path. This frees up network resources and reduces PDD. Several particular embodiments have been described above in which an application server may appropriately determine that it is no longer required in the signaling path and thus remove itself from the signaling path. It would be readily apparent to one skilled in the art that there are many other situations in which the principles of the present invention may be applied. 
     The foregoing Detailed Description is to be understood as being in every respect illustrative and exemplary, but not restrictive, and the scope of the invention disclosed herein is not to be determined from the Detailed Description, but rather from the claims as interpreted according to the full breadth permitted by the patent laws. It is to be understood that the embodiments shown and described herein are only illustrative of the principles of the present invention and that various modifications may be implemented by those skilled in the art without departing from the scope and spirit of the invention. Those skilled in the art could implement various other feature combinations without departing from the scope and spirit of the invention. For example, there are many other situations, in addition to those described herein, in which an application server may determine that it is no longer required in the signaling path and may therefore remove itself from the signaling path.