FIELD OF THE INVENTION
The present invention relates generally to the field of Voice over Internet Protocol (VoIP). More specifically, the present invention relates to the field of bandwidth reduction in a VoIP network.
BACKGROUND OF THE INVENTION
When the underlying voice media of a Contact Center is VoIP, an incoming customer call is received at a gateway. The incoming call is split into a session initiation protocol (SIP) portion and a real time protocol (RTP) portion, sent to a media server (VMS) and a workflow is started for that customer call. The SIP portion of the call contains control signaling to facilitate the routing of the call while the RTP portion of the call contains the actual voice data. In current VoIP networks, when the workflow determines that the customer call is to be on hold, the VMS provides an on-hold playback to the customer through the gateway until the workflow determines that the customer call is to be taken off hold. Because the on-hold playback requires bandwidth for each caller on hold for a continuous amount of time, this system can consume a large amount of bandwidth when many callers are on hold.
The problem with this type of system is apparent and magnified when the VoIP network is connected across large distances through the use of wide area networks (WANs). For example, when a VoIP network has a gateway in a first city, but the customer call received in that gateway is properly routed to a VMS in a second city through a WAN, the on-hold playback must be transmitted continuously through the WAN and the gateway to the caller. Because WAN bandwidth is expensive, and on-hold calls consume bandwidth while the on-hold playback is continuously playing, current systems are unnecessarily expensive and inefficient.
SUMMARY OF THE INVENTION
The present invention is a method and system for reducing bandwidth usage in a voice over internet protocol (VoIP) network that uses holding sounds such as music or announcements. When a customer call enters the network through a gateway, the call is received by a media server (VMS) and a workflow is started. When the workflow determines that the customer call is to be put on hold, the VMS sends an instruction signal to the gateway, instructing the gateway to provide an on-hold music or announcement to the customer call. When the workflow determines that the customer call is to be taken off hold, the VMS sends a second instruction signal to the gateway and the gateway stops the music or announcements and proceeds to further process the call under the direction of the workflow.
One aspect of the present invention includes a method of reducing bandwidth usage in a voice over internet protocol network comprising receiving a customer call in a gateway from a network, routing the customer call to a server and sending an instruction signal from the server to the gateway, wherein the instruction signal prompts the gateway to start or stop an on-hold playback from the gateway to a customer. The method further comprises starting a workflow when the customer call is routed to the server, wherein the workflow determines when the instruction signal is sent to the gateway, an on-hold playback track, an announcement stack and an announcement frequency. The instruction signal is a start playback signal when the customer call is put on hold, further wherein the start playback signal prompts the gateway to start the on-hold playback. The instruction signal is an end playback signal when the customer call is taken off hold, further wherein the end playback signal prompts the gateway to stop the on-hold playback.
The method further comprises saving an interruption point of the on-hold playback when the end playback signal prompts the gateways to stop the on-hold playback, starting a message or a caller prompt after the end playback signal is sent and connecting the customer call to an agent after the end playback signal is sent. When the agent places the customer on hold, the instruction signal is re-sent from the server to the gateway to start the on-hold playback from the gateway to the customer, and the on-hold playback starts from the interruption point. The instruction signal is sent to the gateway over a local connection or over a wide area network. The network is a time division multiplexed network such as a public switched telephone network or the network is a voice over internet protocol network.
Another aspect of the present invention includes a system for reducing bandwidth usage in a voice over internet protocol network comprising means for receiving a customer call from a network, means for routing the customer call to a server and means for instructing the gateway, wherein the instructing means prompts the gateway to start or stop an on-hold playback from the gateway to a customer. The system further comprises means for starting a workflow when the customer call is routed to the server, wherein the workflow determines when the instructing means prompts the gateway, an on-hold playback track, an announcement stack and an announcement frequency. The instructing means sends a start playback signal when the customer call is put on hold, further wherein the start playback signal prompts the gateway to start the on-hold playback. The instructing means sends an end playback signal when the customer call is taken off hold, further wherein the end playback signal prompts the gateway to stop the on-hold playback.
The system further comprises means for saving an interruption point of the on-hold playback when the end playback signal prompts the gateways to stop the on-hold playback, means for starting a message or a caller prompt after the end playback signal is sent and means for connecting the customer call to an agent after the end playback signal is sent. When the agent places the customer on hold, the instructing means prompts the gateway to start the on-hold playback from the gateway to the customer, and the on-hold playback starts from the interruption point. The instructing means prompts the gateway over a local connection or over a wide area network. The network is a time division multiplexed network such as a public switched telephone network or the network is a voice over internet protocol network.
Another aspect of the present invention includes a system for reducing bandwidth usage in a voice over internet protocol network comprising a gateway configured to receive a customer call from a network, and further configured to route the customer call and a server coupled to the gateway, wherein the server is configured to receive the customer call from the gateway, and further configured to send an instruction signal to the gateway, wherein the instruction signal prompts the gateway to start or stop an on-hold playback from the gateway to a customer. The system further comprises a workflow configured to be started when the customer call is routed to the server, wherein the workflow determines when the instruction signal is sent to the gateway, an on-hold playback track, an announcement stack and an announcement frequency.
The instruction signal is a start playback signal when the customer call is put on hold, further wherein the start playback signal prompts the gateway to start the on-hold playback. The instruction signal is an end playback signal when the customer call is taken off hold, further wherein the end playback signal prompts the gateway to stop the on-hold playback. The gateway records an interruption point of the on-hold playback when the end playback signal prompts the gateways to stop the on-hold playback. A message or a caller prompt is started after the end music playback is sent. The customer call is connected to an agent after the end playback signal is sent. When the agent places the customer on hold, the instruction signal is re-sent from the server to the gateway to start the on-hold playback from the gateway to the customer, and the on-hold playback starts from the interruption point. The instruction signal is sent to the gateway over a local connection or a wide area network. The network is a time division multiplexed network such as a public switched telephone network or the network is a voice over internet protocol network.
BRIEF DESCRIPTION OF THE DRAWINGS
FIG. 1 illustrates a graphical representation of an exemplary VoIP network of an embodiment of the present invention.
FIG. 2 illustrates a graphical representation of an exemplary VoIP network of an embodiment of the present invention.
FIG. 3 illustrates a schematic diagram of an embodiment of the present invention.
FIG. 4 illustrates a flow chart representation of a method of an embodiment of the present invention.
DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT
A voice over internet protocol (VoIP) system 100 of an embodiment of the present invention is depicted in FIG. 1. A telephone 120 is utilized by a customer to connect to a VoIP contact center 110 through a public switch telephone network 130 or some other time division multiplexed (TDM) network 140. The customer call is received in the VoIP contact center 110 by a gateway 150. The gateway 150 splits the call into a session initiation protocol (SIP) portion 160 and a real time protocol (RTP) portion 170, sent to a media server (VMS) 180 and a workflow is started for that customer call. The SIP portion 160 of the call contains control signaling to facilitate the routing of the call while the RTP portion 170 of the call contains the actual voice data. The customer call is routed to an appropriate VMS 180 according to a predetermined set of routing criteria.
In the embodiment depicted in FIG. 1, the SIP portion 160 and the RTP portion 170 of the customer call are transmitted along a local transmission line 145 to the VMS 180. Once the customer call reaches the VMS 180, a workflow is started for that customer call. The workflow proceeds through a number of steps that must be completed for that particular customer call. Oftentimes, the workflow will require that the customer call be put on hold. In such a case, the VMS 180 sends an instruction signal through the local transmission line 145 to the gateway 150, instructing the gateway 150 to start an on-hold playback for that customer call. The gateway 150 is capable of starting a continuous pre-recorded or live stream to the telephone 120 through the PSTN 130. The on-hold playback may include music, advertisements or any information the owner of the VoIP contact center wants to relay to the customer while the customer is on hold. It should be noted that the preferred embodiment of the present invention utilizes a VMS 180 to send an instruction to the gateway 150. However, it has been contemplated that any server or other device that is a source of music on hold or controls the application of music on hold may send this instruction as well.
Still referring to FIG. 1, when the workflow is at a stage where the customer call is to be routed to an agent 190, the workflow will prompt the VMS 180 to send an instruction signal through the local transmission line 145 to the gateway 150, instructing the gateway 150 to stop the on-hold playback for that customer call. The VMS 180 will then transfer the customer call to the appropriate agent 190. Likewise, if the workflow requires a customer prompt or other message not in the on-hold playback to be sent to the customer, the workflow will also prompt the VMS 180 to send an instruction signal to the gateway 150 to stop the on-hold playback so that the customer prompt, e.g., “press 1 to continue holding, press 2 to leave a message for us to call you back,” or message can be sent to the customer. After such a customer prompt or message is completed, the workflow will prompt the VMS 180 to send an instruction signal through the local transmission line 145 to the gateway 150, instructing the gateway 150 to start the on-hold playback for that customer call once again. It should be noted that the preferred embodiment of the present invention utilizes explicit instruction signals to start or stop the on-hold playback. However, it has been contemplated that standard VoIP signaling commands from the VMS 180 such as, but not limited to, “no longer listening” or “resuming listening” may be implicitly utilized to start or stop the on-hold playback.
Still referring to FIG. 1, the on-hold playback is selected from a number of on-hold playback tracks. Also, the other messages that may be interspersed during the customer call's on-hold time are each part of an announcement stack. An announcement stack includes a number of related announcements targeted toward a particular customer type. When the workflow prompts the VMS 180 to send the gateway 150 an instruction to start the on-hold playback, the instruction signal will also include instructions from the gateway regarding which on-hold playback track to play, which announcement stack to select and frequency of the announcements. Each announcement in the announcement stack will play at the desired frequency and the announcement stack will continue from the beginning, when the last announcement in the stack is played. Factors that may determine the particular on-hold playback track and announcement stack include, but are not limited to, the type of call, the call origin and caller information.
Still referring to FIG. 1, when the workflow prompts the VMS 180 to send an instruction signal to the gateway 150 to stop the on-hold playback, the gateway 150 will remember the interruption point where the on-hold playback track was stopped. Subsequently, if the agent wishes to put the call back on hold and the instruction signal is sent to start the on-hold playback, the on-hold playback track will re-start at the interruption point. Furthermore, if an announcement is interrupted when the call is connected, upon subsequently being put back on hold, the on-hold playback track will be started at the point the on-hold playback track was interrupted by the announcement. In the preferred embodiment, an interrupted announcement will not be re-started when a call is put back on hold.
The current embodiment of the present invention depicted in FIG. 1 reduces the bandwidth required in systems of the prior art those systems play music from the media server 180, thus consuming large portions of the bandwidth of the local transmission line 145. Furthermore, the local switchgear that control the local transmission lines 145 have packet transfer limitations. By implementing the present invention as described above, those limitations are greatly reduced. It should be understood that alternative embodiments of the present invention may include any network in place of the TDM 140, PSTN 130 and telephone 120, as depicted in FIG. 1. Alternative embodiments may include, but are not limited to, a PBX network, a carrier network or another VoIP network. This should be understood not only in reference to FIG. 1, but in reference to the remaining figures as well.
A VoIP system 100 of the preferred embodiment of the present invention is depicted in FIG. 2. Here, a first contact center 110A and a second contact center 110B are connected by a wide area network (WAN) 200. Such a configuration may occur when a contact center 110A, B has a separate hubs 155A, B in two different cities. The WAN 200 may connect contact centers 110A, B mere miles apart or contact centers 110A, B on the other side of the globe. In any event, WAN 200 bandwidth is expensive and should be conserved in order to operate an efficient contact center 110A, B. It should be understood that additional contact centers 110A,B may be implemented and connected through WAN 200 or local transmission lines 145 (FIG. 1) as required, and that the contact center 110A, B depicted in FIG. 2 is intended to illustrate the preferred embodiment of the present invention. It should also be understood that alternative embodiments may include a VoIP system 100 having a use application other than a contact center 110. Again, alternative embodiments may include a PBX network.
Referring still to FIG. 2, when a customer call originating from a first telephone 120A is received in a first gateway 150A of a first contact center 110A through a first PSTN 130A or other TDM network 140, the first gateway 150A must determine the appropriate VMS 180 to route the customer call according to predetermined routing criteria. While it is not shown, numerous contact centers 110A, B having additional VMS 180 units may be implemented in any given VoIP system 100. When a customer call is routed to a VMS 180 in a second contact center 110B, a WAN 200 is utilized to connect the VMS 180 to the first gateway 150A through a first hub 155A. When the customer call is routed to the VMS 180, a workflow is once again started for that customer call.
The workflow proceeds through a number of steps that must be completed for that particular customer call. Oftentimes, the workflow will require that the customer call be put on hold. In such a case, the VMS 180 sends an instruction signal through the WAN 200 to the first gateway 150A, instructing the first gateway 150A to start an on-hold playback for that customer call. The first gateway 150A is capable to start a continuous pre-recorded or live stream to the first telephone 120A through the first PSTN 130A. Once again, the on-hold playback may include music, advertisements or any information the owner of the VoIP contact center wants to relay to the customer while the customer is on hold.
Still referring to FIG. 2, when the workflow is at a stage where the customer call is to be routed to an agent 190B in the second contact center 110B, the workflow will prompt the VMS 180 to send an instruction signal through the WAN 200 to the first gateway 150A, instructing the first gateway 150A to stop the on-hold playback for that customer call. The VMS 180 will then transfer the customer call to the appropriate agent 190B. Likewise, if the workflow requires a customer prompt or other message not in the on-hold playback to be sent to the customer, the workflow will also prompt the VMS 180 to send an instruction signal to the first gateway 150A to stop the on-hold playback so that the customer prompt, e.g., “press 1 to continue holding, press 2 to leave a message for us to call you back,” or message can be sent to the customer. After such a customer prompt or message is completed, the workflow will determine if continued holding was selected and prompt the VMS 180 to send an instruction signal through the WAN 200 to the first gateway 150A, instructing the first gateway 150A to start the on-hold playback for that customer call once again. Likewise, the workflow will prompt the VMS 180 to send an instruction signal through the WAN 200 to the first gateway 150A, instructing the first gateway 150A to start the on-hold playback for that customer when the agent needs to put the customer on-hold for any reason.
The preferred embodiment of the present invention depicted in FIG. 2 reduces the bandwidth required in systems of the prior art those systems play music from the media server 180, thus consuming large portions of the bandwidth of the WAN 200. The reduction in bandwidth usage can be realized when a VoIP system 100 utilizing the present invention is able to utilize the WAN 200 for relatively short instruction signals, rather than continuous on-hold playbacks to hundreds, or even thousands of customer calls. Furthermore, as stated previously, FIG. 2 is exemplary to illustrate the preferred embodiment of the present invention. Many VoIP systems 100 utilize additional contact centers 110A, B, connected by additional WANs 200, and effectuate a customer call being connected through a number contact centers 110A, B over numerous WANs 200. Thus, the present invention reduces bandwidth usage over all the WANs 200 in a given VoIP system 100 by limiting the WANs to instruction signals, rather than continuous on-hold playbacks. A further advantage of the present invention is that by placing the on-hold playback responsibilities in the gateway 150A, B, advertisements and messages are localized, allowing national as well as local advertising to be relayed to the customer through the on-hold playback.
A schematic block diagram of an embodiment of the present invention is depicted in FIG. 3. Here, the PSTN 130 (or other network) provides the gateway 150 with a customer call 310. The gateway 150, in turn, routes 330 the call to the VMS 180 and a workflow 350 is started for that call. It should be noted that while the workflow 350 may be located in the VMS 180, it can also be located in other parts of the VoIP system such as, but not limited to, the application server. As described previously, instruction signals 340 are sent by the VMS 180 to the gateway 150, according to prompts by the workflow 350. The instructions signals 340 may instruct the gateway 150 to start or stop an on-hold playback 320 to the customer through the PSTN 130.
A method of reducing bandwidth usage 400 of an embodiment of the present invention is depicted in FIG. 4. In step 405, the gateway receives a call from a network. In step 410, the call is routed to an appropriate media server according to predetermined routing criteria. In step 415, a workflow is started for the call. In step 420, the workflow sends a “start playback” instruction signal from the media server to the gateway, thus triggering an on-hold playback from the gateway to the network. In step 425, if the workflow is ready to connect the call to an agent, then in step 450, an “end playback” instruction signal is sent from the media server to the gateway, thereby stopping the on-hold playback from the gateway to the network, and the call is connected to the appropriate agent in step 455. If the workflow is not ready to connect the call in step 425, then the method continues on to step 430.
Still referring to FIG. 4, in step 430, if the workflow does not include any intervening messages or caller prompts, then in step 445, the on-hold playback from the gateway to the network is continued until the workflow is ready to connect the call. The method then proceeds to step 450. If in step 430, the workflow does include intervening messages not in the on-hold playback or caller prompts, then in step 435, an “end playback” instruction signal is sent from the media server to the gateway, thereby stopping the on-hold playback from the gateway to the network. Then, in step 440, the message is played or the caller prompt is started until completion, at which time the “start playback” instruction signal is again sent in step 420.
Still referring to FIG. 4, after the call is connected to the appropriate agent in step 455, the agent may be required to put the call on hold for a variety of reasons, such as to access additional caller information, as a supervisor a question, or even to transfer the call. If the agent does need to place the call on hold, then the method continues back to step 420. If the agent does not need to place the call on hold, then the method ends.
The present invention has been described in terms of specific embodiments incorporating details to facilitate the understanding of the principles of construction and operation of the invention. Such reference herein to specific embodiments and details thereof is not intended to limit the scope of the claims appended hereto. It will be apparent to those skilled in the art that modifications can be made in the embodiment chosen for illustration without departing from the spirit and scope of the invention. Specifically, it will be apparent to one of ordinary skill in the art that the method and system of the present invention could be implemented in several different ways and have several different appearances.
Patent Application
20050249192
