Recording invocation of communication sessions

Systems and methods for recording a communication session between a customer and an agent of a customer center communication system are provided. In this regard, a representative method comprises: configuring a communication component of the customer center communication system to enable recording of a media stream associated with the communication session; transmitting the media stream over a network based on the configuration of the communication component; receiving the media stream over the network; and recording the received media stream.

TECHNICAL FIELD

The present disclosure is generally related to recording media streams associated with interactions between customers and agents.

BACKGROUND

A traditional passive tap recording technique includes recorders that are deployed along routes of communications. In this technique, each recorder operates similar to a “sniffer” by analyzing pass-by communication packets. The recorder records the packets corresponding to certain communication sessions based on its configuration.

In an IP telephony environment at, for example, a customer center, recorders are typically deployed either at a voice gateway, which interfaces between the Internet Protocol (IP) network and the public switched telephone network (PSTN), or at switches in order to stay along the routes of the communications. This technique has the advantages, among others, of (1) minimum intrusion into a communication system, (2) cost effectiveness in deployment for companies with centralized office locations, and (3) easy scalability for compliance recording.

To communicate with any agents at the customer center, a customer communication device, such as a time domain multiplexing (TDM) or a voice over Internet Protocol (VOIP) phone, first sends communication signals to a call-processing device of the customer center, such as a soft switch. The communication signals can be sent either directly to the call-processing device in case of IP to IP communications or via a media processing device, such as a voice gateway in case of TDM to IP. The communication network can be a PSTN network or an IP-based network.

Once the communication signals have been received, the call-processing device then routes the communication signals to an agent phone. After several rounds of communication signal exchange, media communications between the agent's phone and customer's phone can proceed via the media processing device and distribution devices. The distribution devices are network routers and switches.

In order to record the media communications using passive tapping, recorders are deployed at the media processing device or distribution devices using the network traffic monitoring or duplicating features, such as Cisco™ Switch Port Analyzer (SPAN) feature, on these devices. These tapping features are often available to the recorders that are directly connected to the media processing device or distribution devices, namely to recorders deployed a customer center telephony system.

The traditional passive tap recording technique is typically deployed in a telephony environment that has little to no “intelligence” in managing and recording media communications. Customer Centers are perpetually looking for ways to improve the recording features of their telephony components. One way, among others, is to provide more “intelligence” within the customer telephony system and the recording system.

SUMMARY

Systems and methods for recording a communication session between a customer and an agent of a customer center communication system are provided. In this regard, a representative method comprises: configuring a communication component of the customer center communication system to enable recording of a media stream associated with the communication session; transmitting the media stream over a network based on the configuration of the communication component; receiving the media stream over the network; and recording the received media stream.

DETAILED DESCRIPTION

Customer center includes, but is not limited to, outsourced contact centers; outsourced customer relationship management, customer relationship management, voice of the customer, customer interaction, contact center, multi-media contact center, remote office, distributed enterprise, work-at-home agents, remote agents, branch office, back office, performance optimization, workforce optimization, hosted contact centers, and speech analytics, for example.

Additionally, included in this disclosure are embodiments of integrated workforce optimization platforms, as discussed in U.S. application Ser. No. 11/359,356, filed on Feb. 22, 2006, entitled “Systems and Methods for Workforce Optimization,” which is hereby incorporated by reference in its entirety. At least one embodiment of an integrated workforce optimization platform integrates: (1) Quality Monitoring/Call Recording—voice of the customer; the complete customer experience across multimedia touch points; (2) Workforce Management—strategic forecasting and scheduling that drives efficiency and adherence, aids in planning, and helps facilitate optimum staffing and service levels; (3) Performance Management—key performance indicators (KPIs) and scorecards that analyze and help identify synergies, opportunities and improvement areas; (4) e-Learning—training, new information and protocol disseminated to staff, leveraging best practice customer interactions and delivering learning to support development; and/or (5) Analytics—deliver insights from customer interactions to drive business performance. By way of example, the integrated workforce optimization process and system can include planning and establishing goals—from both an enterprise and center perspective—to ensure alignment and objectives that complement and support one another. Such planning may be complemented with forecasting and scheduling of the workforce to ensure optimum service levels. Recording and measuring performance may also be utilized, leveraging quality monitoring/call recording to assess service quality and the customer experience.

Exemplary systems are first discussed with reference to the figures. Although these systems are described in detail, they are provided for purposes of illustration only and various modifications are feasible. After the exemplary systems are described, examples of flow diagrams of the systems are provided to explain the manner in which media streams associated with an interaction can be recorded.

Systematic Diagram Using a Proxy Server

Referring now in more detail to the figures,FIG. 1is a schematic diagram of an embodiment of a system in which media communications can be recorded using a proxy server. A customer center telephony system105has one or more agent phones110,115,120that are coupled to one or more computer-telephone integration (CTI) servers125and one or more call control servers130. The customer center telephony system105is coupled to the recording system133, which includes one or more proxy servers135, one or more recorders140,145,150, and one or more recording control servers155.

In general, the proxy server135is a computer that offers a computer network service to allow clients to make indirect network connections to other network services. A client connects to the proxy server135, then requests a connection, file, or other resource available on a different server, e.g., the CTI server125, call control server130, and recording control server155, among others.

The proxy server135functions as a single node for the customer center telephony system105or any third party system (not shown) to communicate with. The proxy server can pass requests (modified) from the customer center telephony system105on to the recorders140,145,150to complete the requests.

Alternatively or additionally, the proxy server accumulates and saves files that are most often requested in a cache. The proxy server135provides the resource either by connecting to the specified server or by serving it from the cache. In some cases, the proxy server135can alter the client's request or the server's response for various purposes. The proxy server135can also serve as a firewall. In one embodiment, the proxy server135can be described as a buffer between a computer and the resources being accessed.

The cache of the proxy server135may already contain information needed by the time of the request, making it possible for the proxy server135to deliver the information immediately. Therefore, the proxy server135can potentially increase the speed of communication to the servers.

Alternatively or additionally, the proxy server135can operate in a hybrid recording environment and communicate with recorders that operate in a hybrid recording mode, in which both active recording and passive sniffing are performed.

It should be noted that the proxy server135is shown inFIG. 1as a separate device from the other components of the recording system133. However, the proxy server135can also be deployed on one of the recorders140,145,150or on other servers, such as the recording control servers155. The proxy server135can communicate with the CTI server125and/or the call control server130to receive call events and other call related information such as agent identifications. The communication can also involve the recording of at least one media stream associated with an interaction within the customer center telephony system105.

The customer center telephony system105can be configured to send call requests to or receive call requests from the proxy server135. The customer center telephony system105can further be configured to send duplicated media streams associated with the interactions to be recorded to the destinations negotiated in the call request process. The proxy server135can also invoke and send requests to the customer center telephony system105to request the duplicated media streams of interactions based on received CTI events and information or other call notification systems, such as SIP, for example.

The proxy server135can route the duplicated media streams associated with the interactions to the recorders140,145,150based on various conditions, criteria, or policies. The proxy server135can also re-direct the calls to a different proxy server based on load, bandwidth, or other configurations. The proxy server135can communicate with the recorders140,145,150using standard call control protocols, such as session initiation protocol (SIP), H323, media gateway control protocol (MGCP), or skinny client control protocol (SCCP), or using proprietary protocols, or using static configurations. The operation of the proxy server is further described in relation toFIGS. 6-8.

Session Initiation Protocol (SIP)

Briefly, SIP is an application layer control simple signaling protocol for Voice Over Internet Protocol (VoIP) implementations. SIP is a textual client-server based protocol that the end user systems and proxy servers can provide call forwarding, callee and caller number identification, basic Automatic Call Distribution (ACD) and personal mobility, among others.

SIP addresses are generally in the form of a Uniform Resource Location (URL). The SIP addresses can be embedded in Web pages and therefore can be integrated as part of implementations such as Click to talk, for example. SIP using simple protocol structure, provides fast operation, flexibility, scalability and multiservice support. SIP provides its own reliability mechanism. SIP creates, modifies and terminates sessions with one or more participants. These sessions include Internet multimedia conferences, Internet telephone calls and multimedia distribution. Members in a session can communicate using multicast or using a mesh of unicast relations, or a combination of these. SIP invitations used to create sessions carry session descriptions, which allow participants to agree on a set of compatible media types.

SIP supports user mobility by proxying and redirecting requests to the user's current location. Users can register their current location. SIP is not tied to any particular conference control protocol. It is designed to be independent of the lower-layer transport protocol and can be extended with additional capabilities. SIP transparently supports name mapping and redirection services, allowing the implementation of Integrated Services Digital Network (ISDN) and Intelligent Network telephony subscriber services. These facilities also enable personal mobility which is based on the use of a unique personal identity. SIP supports five facets of establishing and terminating multimedia communications: 1) User location, 2) User capabilities, 3) User availability, 4) Call setup, and 5) Call handling.

SIP can also initiate multi-party calls using a multipoint control unit (MCU) or fully-meshed interconnection instead of multicast. Internet telephony gateways that connect Public Switched Telephone Network (PSTN) parties can also use SIP to set up calls between them. SIP is designed as part of a multimedia data and control architecture currently incorporating protocols, such as Resource Reservation Protocol (RSVP), Real-Time Transport Protocol (RTP), Real-Time Streaming Protocol (RTSP), and Service Advertising Protocol (SAP), among others. However, the functionality and operation of SIP does not depend on any of these protocols. SIP can also be used in conjunction with other call setup and signaling protocols. In that mode, an end system uses SIP exchanges to determine the appropriate end system address and protocol from a given address that is protocol-independent. For example, SIP could be used to determine that the party can be reached using H.323, to find the H.245 gateway and user address, and then use H.225.0 to establish the call.

H.323 is a protocol that generally provides audio-visual communication sessions on any packet network. Currently, H.323 is implemented by various Internet real-time applications such as NetMeeting and Ekiga. H.323 is commonly used in voice over Internet Protocol (VoIP, Internet Telephony, or IP Telephony) and IP-based videoconferencing.

Media Gateway Control Protocol (MGCP)

MGCP is a standard protocol generally for handling the signaling and session management needed during a multimedia conference. The protocol defines a means of communication between a media gateway, which converts data from the format required for a circuit-switched network to that required for a packet-switched network and the media gateway controller. MGCP can be used to set up, maintain, and terminate calls between multiple endpoints. MGCP can be used for controlling telephony gateways from external call control elements, such as the media gateway controllers or call agents. MGCP assumes a call control architecture where the call control intelligence is outside the gateways and handled by external call control elements. The MGCP assumes that these call control elements, or Call Agents, can synchronize with each other to send coherent commands to the gateways under their control. MGCP is, in essence, a master/slave protocol, where the gateways are generally expected to execute commands sent by the Call Agents.

Skinny Client Control Protocol (SCCP)

SCCP defines a simple and easy to use architecture. An H.323 proxy server can be used to communicate with the Skinny Client using the SCCP. In such a case, the telephone is a skinny client over IP, in the context of H.323. A proxy server is used for the H.225 and H.245 signaling. The end stations (such as the agent phone110,115,120) can run what is called the Skinny Client, which consumes less processing overhead. The Skinny Client can communicate with the call control server130to establish a call with another phone110,115,120. Once the call control server130has established the call, the two phones use connectionless (UDP/IP-based) communication for audio transmissions. Costs and overhead are thus reduced by confining the complexities of H.323 call setup to the call control server130, and using the Skinny protocol for the actual audio communication into and out of the end stations.

Simply Call Flow Diagram

FIG. 2illustrates an exemplary call flow in recording media streams associated with a call. Upon receiving a call request for an agent's extension via line210, a call control server230checks the recording configuration for that extension. If the extension is configured for recording, in addition to the normal call setup procedures, the call control server230sends call requests to a proxy server235and the agent phone220to establish dialogs for recording via lines225and237. Upon receiving the call request from the call control server230, the proxy server235routes the call from the call control server230to one of the recorders240,245via lines237,242. The recorders are registered with the proxy server235based on, for example, a recording load balancing algorithm, among others. Upon receiving a positive confirmation from the selected recorder245, the proxy server235can include an IP address and port of the selected recorder245in its responses as part of the call control flow.

Upon having successfully established the dialogs, the call control server230instructs the agent's phone220via line225to send duplicated media streams of the on-going call to the recorder245. The agent phone220duplicates media streams and sends the duplicated media streams via line247, in the format of the, for example, real-time transport protocol (RTP) packets, of the call directly to the recorder245. The recorder245can associate two of the media streams of the same call and record them as, for example, a single media stream. When the call ends, the call control server230terminates the dialog with the proxy server235using the call control protocol via line237. The proxy server235sends a call end message to the recorder242via line242.

Call Flows for Recording

Using a Proxy Server

FIGS. 3A-Cillustrate exemplary call flows for recording one or more duplicated media streams associated with a call using a proxy server, such as that shown inFIG. 1. The call control servers330A-C send call requests to proxy servers335A-C and the agent phones320A-C to establish dialogs for recording media streams via lines337A-C. The two or more duplicated media streams can be sent to a recording system in a single call dialog or as two separate call dialogs. In case of two separate call dialogs, a unique call identification of the original call may be included in the information of the two call dialogs to link the relationship between them.

Referring now toFIG. 3A, the proxy server335A instructs the call control server330A via line337A to re-direct the calls to a recorder345A. The call control server330A instructs the agent phone320A, gateways (not shown), conference bridges (not shown) and other telephony components to duplicate the media streams and transmit the duplicated media streams to the recorder345A. The call control server330A communicates with the recorder345A to establish a dialog for recording the duplicated media streams via line339. The agent phone320A transmits the duplicated media streams to the recorder345A via line341.

Referring toFIG. 3B, the proxy server335B proxies the call and directs an agent phone320B to send the duplicated media streams directly to recorder345B. The proxy server335B communicates with the recorder345B to establish a dialog for recording the duplicated media streams via line342. The call control server330B instructs with the agent phone320B to duplicate the media streams and transmit the duplicated media streams directly to the recorder345B. The agent phone320B transmits the duplicated media streams directly to the recorder345B via line344.

Referring toFIG. 3C, the proxy server335C proxies the call as well as the duplicated media streams. The proxy server345C communicates with the recorder335C to establish a dialog for recording the duplicated media streams via line346. The call control server330C instructs the agent phone320C to duplicate the media streams and transmit the duplicated media streams to the proxy server335C. The agent phone320C transmits the duplicated media streams to the proxy server335C via line348. The proxy server335C receives the duplicated media streams and transmits the duplicated media streams to the recorder345C via line352.

The proxy servers335inFIGS. 3A-Cmay use the standard call control calls such as SIP, H323, which may or may not be the same as the protocol it uses to communicate with the telephony system, to communicate with the recorders. The proxy server may also use other protocols to communicate with the recorders345.

Recorder Failover Solution

FIG. 4is a schematic diagram that illustrates an embodiment that utilizes recorder failover. A proxy server435communicates with a customer center telephony system405, which includes an agent phone420and a call control server430, among others. The proxy server435communicates with multiple recorders440,445of a recording system433. If the recorder440malfunctions, the proxy server435can communicate with the recorder445to record media streams associated with the interactions within the customer center telephony system405, and vice versa. If the recorders440,445malfunction, the proxy server435can communicate with another recorder (not shown) to record the media streams associated with the interactions within the customer center telephony system405. Alternatively or additionally, the proxy server435detects that a recorder fails and sends a redirect message (or reINVITE message as in the case with SIP) to the Customer Center telephony system to transmit the existing call/recordings to a “good” recorder.

Proxy Failover Solution

FIG. 5is an exemplary schematic diagram that illustrates a embodiment that utilizes proxy failover. The proxy failover solution can use a link protector534, which communicates with a customer center telephony system505. The system505includes an agent phone520and a call control server530, among others. Multiple proxy servers connect to the link protector534. In this example, proxy server535serves as a primary server and proxy server537and other servers (not shown) serve as back-ups. The proxy servers535,537can have the same IP address on the network interface, connecting to the link protector. These proxy servers535,537connect to the same pool of recorders540,545. During normal operation, the link protector534passes network traffic to the primary proxy server535. Upon having detected that connection to the primary proxy server535is down, the link protector534passes traffic to the backup proxy server357.

Alternatively or additionally, proxy failover can be achieved by configuring the multiple proxy servers535,537with different Time IP addresses but the same domain name system (DNS) name. In this implementation, the multiple proxy servers535,537, connecting to the same pool of recorders are utilized. These proxy servers535,537, with different IP addresses on the interface connecting to the customer center telephony system505, are registered with the same DNS name. The customer center telephony system505is configured with the DNS names of the proxy servers535,537for communicating with the proxy servers535,537. For example, the customer center telephony system505queries for the DNS of the proxy servers535,537to obtain the proxy servers' IP addresses. Responsive to the query, the customer center telephony system505receives the proxy servers' IP addresses and maintains a list of the IP addresses in a DNS database. The customer center telephony system505generally exhausts the entire list to find an IP address to successfully connect to the proper proxy server.

Alternatively or additionally, proxy failover can be achieved by configuring a primary and multiple secondary proxy servers in the customer center telephony system505. Alternatively or additionally, a failover solution can be achieved by instructing the recorders540,545to switch to passive (sniffing) recording provided that the recorders540,545are configured with a hybrid recording mode.

Flow Diagram Depicting Operation of Recording Media Streams

FIG. 6is a flow diagram that illustrates operation of an embodiment in which media streams associated with an interaction are recorded. Beginning with block605, a customer center telephony system communicates with a proxy server for recording the media streams of the interaction. For example, a customer center's telephony infrastructure can configure each telephony component, such as an agent phone with an extension, gateway, call control server, and CTI server, among others, with recording options. The recording options can include, but are not limited to the following:1. “Do not record”, which indicates calls on an extension should never been recorded.2. “100% record”, which indicates that all calls on the extension should be recorded.3. “Application driven record”, which means the telephony components receive instructions from applications of, for example, a recording system or a policy system, to initiate a recording session of a call on the extension.4. Record communication sessions based on business rules of the communication system, the business rules including recording all communication sessions in a queue or at a predetermined time.

In block610, the proxy server stores recording information received from the customer center telephony system or the recording subsystem, among others. The recording information can include the recording options as mentioned above for initiating a recording session.

In block615, the proxy server communicates with the customer center telephony system or the recording subsystem and determines whether to record the media streams associated with the interaction based on the communication. In block620, responsive to the determination to record the media streams, the proxy server provides recording instructions to the customer center telephony system and the recording system based on the recording information.

In the first two options mentioned above, the customer center telephony system receives the calls and determines whether to invoke the calls to the proxy server for recording. In the “application driven record” option, the customer center telephony system waits to receive instructions from the recording system and a policy system. For example, the recording system may get the events of the calls on the agent phones via a CTI link. The instructions for recording from the recording system can be delivered via the CTI link to the customer center telephony system. In turn, the customer center telephony system can invoke the call recording session.

Alternatively or additionally, the instructions for recording can be sent to the customer center telephony system in the form of the call control protocol to directly invoke the call recording session. Alternatively or additionally, the customer center's telephony infrastructure may choose to set a recording option to all extensions as a default option. For example, the customer center telephony system can set “application driven record” as the recording option without any configuration involved. In this case, the proxy server invokes the requests for recording the media streams.

In a recording system where a hybrid recording mode is provided, the proxy server can decide if it needs to instruct the telephony system for active recording based on whether the call is already being recorded using the sniffing method.

In block625, the media streams associated with the interaction within the customer center telephony system are associated with each other so that, for example, the media streams can be stitched together as a single recording for archival and replay. In block630, the proxy server or the customer center telephony system initiates a recording session with the recording system. In block635, the recording system receives and records the media streams based on the recording instructions from the proxy server.

Flow Diagram Depicting Operation of a Proxy Server

FIG. 7is a flow diagram that illustrates an exemplary operation for recording an interaction between a customer and an agent using a proxy server, such as that shown inFIG. 1. In block705, the proxy server communicates with a customer center telephony system. The communication is related to the recordation of the interaction between the customer and the agent. In block710, the proxy server stores data that includes recording criteria corresponding to the recordation of the media streams associated with the interaction. The data is generally saved in the cache of the proxy server.

In block715, the proxy server determines whether to record the media streams associated with the interaction based on the communication between the customer center telephony system and the proxy server. For example, the customer center telephony system can invoke the proxy server to record the interaction. The proxy server receives information pertaining to the invocation from the customer center telephony system and facilitates the process for recording the interaction.

In block720, responsive to the determination of recording the media streams, the proxy server receives information about the media streams to facilitate associating the media streams with each other. In block725, the proxy server transmits instructions to the customer center telephony system for initiating a recording session. The instructions include duplicating the media streams and transmitting the duplicated media streams to a recording subsystem of the recording system. Alternatively or additionally, the instructions can include transmitting the duplicated media streams to the proxy server, which relays the duplicated media streams to the recording subsystem. Alternatively or additionally, the proxy server can provide instructions to the recording system to initiate the recording session.

In block727, the proxy server can intelligently route requests to the recorders based on various criteria, policies, or recorder conditions. Alternatively, the customer center telephony system can also intelligently route requests to the recorders based on various criteria, policies, or recorder conditions. In this regard, either the proxy server or the customer center telephony system, or both, can:(1) Balance call load by routing calls to the recorders based on the information of the recorders, such as current recording load and CPU usage, among others;(2) Route calls to recorders based on information in the call control protocols (e.g., tie two streams associated with a single call to the same recorder) or other policies such as co-residency of media, co-residency of media includes, but is not limited to, voice and screen capture, among others;(3) Route calls to the recorders based on a network topology and costs;(4) Redirect calls to different proxy servers based on the network topology and the recording load; and(5) Reject the call if hybrid recording mode is configured on the recorders and one of the recorder is already recording the call via passive method; and(6) Rejecting the media stream based on runtime or configuration criteria, the criteria including the recorder being over-loaded.

In block730, the recording subsystem receives and records the media streams based on the associated information. The information to associate the media streams can be passed to the recording system via the call control protocol, such as SIP or H323, or via the CTI link. The recording system can use the information associated with the call to stitch the two media streams and record them as a single media stream. For example, a left media stream can be associated with a right media stream by using a unique identification stored in the left and right media streams. The unique identification corresponds to the interaction.

Alternatively or additionally, the left and right media streams are transmitted to first and second recording subsystems of the recording system, respectively. The first and second recording subsystems provide the left and right media streams, respectively, to be stitched together as a single stream at, for example, archival and replay systems. Alternatively or additionally, the left and right media streams can be transmitted from the customer center telephony system to the archival and/or replay systems, which both can stitched together the left and right media streams as a single stream.

With regard to stitching, some embodiments can, for example, stitch the segments together with no gap at replay, stitch the segments together with a gap at replay, stitch the segments together with alternative content at replay (such as silence or other identifiers to signify that secured information has not been recorded and/or deleted and/or is protected from replay). For example, such a system may also provide indication between the stitched sides, which can be, for example, two or more sides that would provide indication to a user that something was removed for security purposes.

Network Topology

A network topology is the pattern of links connecting pairs of nodes of a network. The network topology provides the “shape” of a local area network (LAN) or other communications system. A given node has one or more links to others, and the links can appear in a variety of different shapes. The simplest connection is a one-way link between two devices. A second return link can be added for two-way communication. Modern communications cables usually include more than one wire in order to facilitate this, although very simple bus-based networks have two-way communication on a single wire.

One way, among others, to add more computers into a network is by daisy-chaining, or connecting each computer in series to the next. If a message is intended for a computer partway down the line, each system bounces the message along in sequence until the message reaches the destination. A daisy-chained network can take two basic forms: linear and ring.

A linear topology puts a two-way link between one computer and the next. By connecting the computers at each end, a ring topology can be formed. An advantage of the ring is that the number of transmitters and receivers can be cut in half, since a message will eventually loop all of the way around. When a node sends a message, each computer in the ring processes the message. If a computer is not the destination node, the computer passes the message to the next node, until the message arrives at its destination. If the message is not accepted by any node on the network, the message travels around the entire ring and returns to the sender. This potentially results in a doubling of travel time for data, but since the message is traveling at a significant fraction of the speed of light, the loss is usually negligible.

Another network topology, among others, is a star topology, which reduces the chance of network failure by connecting all of the systems to a central node. When applied to a bus-based network, this central hub rebroadcasts the transmissions received from any peripheral node to the peripheral nodes on the network, sometimes including the originating node. The peripheral nodes may thus communicate with all others by transmitting to, and receiving from, the central node only. The failure of a transmission line linking the peripheral nodes to the central node can result in the isolation of that peripheral node from all others, but the rest of the system remains unaffected.

Another network topology, among others, is a tree topology, which can be viewed as a collection of star networks arranged in a hierarchy. The tree topology has individual peripheral nodes (e.g., leaves) which are used to transmit to and receive from one other node only and may not act as repeaters or regenerators. Unlike the star network, the function of the central node may be distributed.

As in the conventional star network, individual nodes may thus still be isolated from the network by a single-point failure of a transmission path to the node. If a link connecting a leaf fails, that leaf is isolated; if a connection to a non-leaf node fails, an entire section of the network becomes isolated from the rest.

In order to alleviate the amount of network traffic that comes from broadcasting everything everywhere, more advanced central nodes were developed that would keep track of the identities of different systems connected to the network. These network switches can “learn” the layout of the network by first broadcasting data packets everywhere, then observing where response packets come from.

In a mesh topology, there are at least two nodes with two or more paths between them. A special kind of mesh, limiting the number of hops between two nodes, is a hypercube. The number of arbitrary forks in mesh networks makes them more difficult to design and implement, but their decentralized nature makes them very useful. This is similar in some ways to a grid network, where a linear or ring topology is used to connect systems in multiple directions. A multi-dimensional ring has a toroidal topology, for instance.

A fully connected, complete topology or full mesh topology is a network topology in which there is a direct link between all pairs of nodes. In a fully connected network with n nodes, there are n(n−1)/2 direct links. Networks designed with this topology are usually very expensive to set up, but have a high amount of reliability due to multiple paths data can travel on.

Hybrid networks use a combination of any two or more topologies in such a way that the resulting network does not have one of the standard forms. For example, a tree network connected to a tree network is still a tree network, but two star networks connected together (known as extended star) exhibit hybrid network topologies. A hybrid topology is always produced when two different basic network topologies are connected. Two common examples for Hybrid network are: star ring network and star bus network. The Star ring network consists of two or more star topologies connected using a multistation access unit (MAU) as a centralized hub. The Star Bus network consists of two or more star topologies connected using a bus trunk (the bus trunk serves as the network's backbone).

While grid networks have found popularity in high-performance computing applications, some systems have used genetic algorithms to design custom networks that have the fewest possible hops in between different nodes. Some of the resulting layouts are nearly incomprehensible, although they do function quite well.

Flow Diagram Depicting Operation of a Proxy Server in a Secure Environment

FIG. 8is a flow diagram that illustrates operation of an embodiment in which a proxy server facilitates recording of encrypted media streams from the customer center telephony system. In the case where encryption is used in the customer center telephony system, duplicated media streams delivered to the recording system can be encrypted. The encrypted media streams have electronic keys for decryption of the media streams. In block805, the electronic keys can be delivered to the proxy server from a telephony component such as a phone, which sends the encrypted duplicated media streams, in a secure connection. The secure connection may be achieved via a secure version of the call control protocols such as secure session initiation protocol (SIP) using transport layer security (TLS) or via a secure CTI link.

Upon detecting encryption and receiving the electronic keys, the proxy server determines how to decrypt the encrypted media streams, shown in block810. The proxy server may choose to directly forward the keys to the selected recorder via a secure connection, shown in block815. Again the secure connection may be a secure version of standard call control protocols such as secure sockets layer (SSL) and transport layer security (TLS) or proprietary socket connection secured by SSL or TLS.

In the scenario where the proxy server chooses to proxy the media streams, the proxy server directs the customer center telephony system to send the encrypted duplicated media streams to itself, shown in block820. The proxy server decrypts the encrypted media streams using the electronic keys passed from the customer center telephony system. The proxy server forwards the decrypted media streams to the selected recorders via a secure connection between the recorder and itself. In block825, the recording subsystem can receive and record the media streams.

Secure Sockets Layer (SSL) and Transport Layer Security (TLS)

Secure Sockets Layer (SSL) and Transport Layer Security (TLS), its successor, are cryptographic protocols which provide secure communications on the Internet for such things as e-mail, internet faxing, and other data transfers. There are slight differences between SSL and TLS, but the protocol remains substantially the same. The term “SSL” as used here applies to both protocols unless clarified by context.

SSL provides endpoint authentication and communications privacy over the Internet using cryptography. In typical use, only the server is authenticated (e.g., its identity is ensured) while the client remains unauthenticated; mutual authentication requires public key infrastructure (PKI) deployment to clients. The protocols allow client/server applications to communicate in a way designed to prevent eavesdropping, tampering, and message forgery.

SSL involves three basic phases: 1) Peer negotiation for algorithm support, 2) Public key encryption-based key exchange and certificate-based authentication, and 3) Symmetric cipher-based traffic encryption. The SSL protocol exchanges records; each record can be optionally compressed, encrypted and packed with a message authentication code (MAC). Each record has a content_type field that specifies which protocol is being used.

In general, the client sends and receives several handshake structures as follows. The client sends a ClientHello message specifying the list of cipher suites, compression methods and the highest protocol version it supports. Then the client receives a ServerHello, in which the server chooses the connection parameters from the choices offered by the client earlier.

When the connection parameters are known, client and server exchange certificates (depending on the selected public key cipher). The server can request a certificate from the client, so that the connection can be mutually authenticated. Client and server negotiate a common secret called “master secret”, or simply encrypting a secret with a public key that is decrypted with the peer's private key. Other key data is derived from this “master secret” (and the client and server generated random values), which is passed through a carefully designed “Pseudo Random Function”.

TLS/SSL have a variety of security measures. Some security measures, among others, are numbering all the records and using the sequence number in the MACs and using a message digest enhanced with a key (so only with the key can you check the MAC).

TLS/SSL protect against several “attacks,” like those involving a downgrade of the protocol to previous (less secure) versions, or weaker cipher suites. TLS/SSL include a hash algorithm in which the message that ends the handshake (“Finished”) sends a hash of the exchanged data seen by both parties.

TLS/SSL further include a pseudo random function that splits the input data in two (2) halves and processes them with different hashing algorithms (e.g., Message-Digest algorithm 5 (MD5) and Secure Hash Algorithm (SHA)). Then the pseudo random function logically processes (e.g., XOR) the two (2) halves together. This way the data protects itself in the event that one of these algorithms is found vulnerable.