Showing posts with label Voice over IP. Show all posts
Showing posts with label Voice over IP. Show all posts

IP Call Center Technology - Call Monitoring and Recording, Trunk Side. by Hoang Lihuo


    Analyzing all of the external interactions with customers, suppliers and other third parties is an important part of IP call center technology.

    Every business will seek to gain a competitive edge from better understanding their customers and acting on insights contained in interactions with them across all channels.

    In addition, agent performance, liability concerns and threat management are matters that require some form of call monitoring and recording for all contact centers.

    There are issues of how this can be technically accomplished in an IP telephony environment.

    Legacy telephone systems had a fairly easy mechanism with which to work. Calls were continuous streams of analog or digital voice, either on the trunk side or the line side of a switch.

    These streams could be recorded either by physically tapping a circuit or port or copying a TDM channel within the switch.

    However, voice over IP transmission is performed using independent voice packets representing snippets of a call interspersed with voice packets from other calls, data packets and signaling packets.

    A physical tap would result in a collection of garbage.

    A few vendors have voice over IP recording systems and they use some technical tricks to monitor and record calls.

    The components involved are call control information translation software, call content recorders, and management and administrative systems. 

    One or more recorders are placed in the contact center network depending upon the size of the network and types of calls that are to be monitored.

    The recorders are programmed to monitor or record certain calls; however, the recorders need to know information about the call packets, the Real Time Protocol packets, that would include end-point addressing and type information at a minimum.

    This signaling level data could come from a CTI server, from direct CTI output from a call control server (this will eventually be replaced by a data application-to-application interface), or it may be gleaned from the call set-up protocol packets, such as SIP packets, as they go by the recorder.

    Once the proper packets are identified for recording, copies are sniffed or snooped (sucked or plucked if you will) off the network into storage.

    Markers and labels are added to the recorded content for later identification, retrieval, or analysis by other programs which are part of the recording system.

    There are several types of monitoring and recording mechanisms, as a way of classifying the activity.

    Passive recording means that automated procedures or programmed procedures are in place to record either all of the calls, bulk recording, or certain calls, selective recording, on a link or part of a network.

    Active recording means a person must initiate the recording process, say an agent who wants to record a threatening or abusive conversation at the moment.

    In addition to these formal types or recoding there are two other variations to mention. One is the tagging, as it is called, of additional information to the recorded call for use in the analysis phase. Tagged information may include things like the application being used at the time of the call or ANI data of the call.

    Recording of calls can also be accomplished in the service provider's network as part of a hosted contact center package.

    This diagram shows one of the common configurations used to record calls.         Incoming and outgoing calls can be recorded by placing the recorder on the trunk side of the IP telephony system.

    Calls relating to the PSTN will traverse a gateway. Calls relating to an IP WAN will traverse an edge router. The recorder will have access to all external calls sitting logically between the ingress gateway or router and the rest of the telephony system – before the calls are shuttled to other parts of the network.

    Physically, this may be accomplished by having the recorder connect to the network through a dumb, three-port hub, which broadcasts all packets to all ports including the recorder. The recorder would need a "promiscuous" network interface card to receive all such packets.

    The recorder is programmed to record all or certain calls whose packets can be identified with Computer-Telephony Interface (CTI) information emanating from the call control server or from the call set-up/transfer signaling (SIP for example) going to and from the gateway or router. The recorder is receiving all traffic through the hub.

    The trunk side attached recorder option is a straight forward solution and can capture the external calls at a good aggregation point in the network. This could include self-service and information-gathering calls that go to the IVR before being passed to an agent.

    This technique is not intrusive and does not consume network bandwidth for the recording process.  If CTI information is not available to the recorder, rich information for tagging will not be available. However, internal calls, such as agent to agent, agent to supervisor, etc. will be missed.


VoIP System Components. by Hoang Lihuo


A VoIP system includes the following basic elements:

    • terminals, including IP phones and software applications running on computers;
    • LAN infrastructure, consisting of wiring and switches;
    • a softswitch or call manager, running a SIP server that acts as a proxy to set up phone calls, manages the terminals, provides registration, authentication and permission management of the phones, plus software applications like call detail record generation, Interactive Voice Response (IVR) and Automated Call Distributor (ACD) functions;
    • a router, which connects LAN broadcast domains together and to WAN circuits;
    • gateways to perform format conversions between the VoIP world and the PSTN;
    • and optional connections to the PSTN, the Internet and to managed IP VPN services and SIP trunking services.
    • A firewall is needed when connecting to the Internet. 

VoIP Components

  • IP phone – In order to send and receive voice calls, IP phones make use of a type of network connection known as Ethernet network connection. An example of IP telephony device is Cisco IP and Cisco 7975G phone on the first picture up here.
  • Gateway – A gateway is able to promote calls which take place between various networks. With the help of a gateway you can place a call between your IP phone and your office. You could also place a call to the PSTN to call your home.
  • Call agent – Most of the characteristics which were formerly a part of PBX have now been replaced by Call Agents. For instance, to approximately conclude how calls are routed, a call agent can be configured with the help of rules. An example of such a call agent is Cisco UCM (Unified Communication Manager)
  • Application server – Application Servers provide on the top services such as voice mail in a VoIP environment.
  • Gatekeeper – Gatekeepers can be metaphorically regarded as the traffic police of the WAN. The bandwidth in WAN network is usually not widely available and so, a gatekeeper can keep an eye on the amount of bandwidth which is accessible. And so, if such a situation occurs that the amount of bandwidth is not able to sustain another voice call, then the gatekeeper can reject all trials of such calls in the future.
  • Videoconference station – Videoconference stations are devices and/or software (such as Cisco Unified Video Advantage) that allow a calling or called party to view and/or transmit video as part of their telephone conversation.
  • Multipoint Control Unit MCUs have been proved to be helpful when it comes to conference calling. Several people are talking together at the same time during a conference call. And everyone can listen to them. In order to mix these audio streams, it takes processing power to do so. Such a source of processing power is the MCU. MCU can contain DSP (Digital sound processor) which are devoted to the circuitry of the computer that is able to mix these audio streams.
  • Voice-enabled switch – A Cisco mechanism switch with voice feature is able to facilitate aligned power to a connected Cisco IP phone, thus there is no need for any outside power source connected to the IP Phone. And a switch with voice feature is able to distinguish between the voice frames which come from the connected IP phone and send those frames to advanced precedence then other frames.

Basic System Architecture by Hoang Lihuo

    System architecture is the conceptual model that defines the structure, behavior, and more views of a system. An architecture description is a formal description and representation of a system, organized in a way that supports reasoning about the structures and behaviors of the system.

System architecture can consist of system components and the sub-systems developed, that will work together to implement the overall system. There have been efforts to formalize languages to describe system architecture, collectively these are called Architecture description languages (ADLs).

    Devices, such as computers, servers, printers, etc., that connect to a local area network (LAN) are normally called stations or terminals . We use both in this course. A wireless access point (AP) is a device connected to a LAN to allow wireless stations to become part of the LAN. It is a transceiver. In a simple case, the AP may only serve one station. Normally, the AP will serve more than one station. In addition, there may be more than one AP connected to a LAN. This provides additional capacity and can serve to cover a large geographic area by placing.

    APs covering the area. APs can stand alone and serve to connect only wireless devices to each other. Normally, however, they are used in conjunction with a wired LAN. A primary use is connection to the public Internet or to a large corporate enterprise network and somewhere these networks get back to wires.

    The range of operations is measured from the AP. For example, the range of 802.11b for 11 Mb/s is about 100 feet. This distance is measured from the AP in all directions and is actually a sphere although most people only think of it as a horizontal circle around the AP. But the AP is a radio transmitter (and receiver) and the transmissions radiate in all directions from the antenna. If the AP is located on the fifth floor of a building, it will radiate around the fifth floor but also radiate up to the sixth and seventh floors as well as down to the fourth and third floors. The penetration of the radio waves through the floors may not be as great as through the walls and the sphere of radiation may be somewhat “squashed” but some radiation up and down will occur. Since most wireless applications connect to a wired LAN or to some other network like the Internet, APs also come integrated into other LAN devices such as bridges or routers.

    Multiple access points covering a geographic area such as a campus or large office complex allow roaming across the area. User stations such as laptop computers with wireless network interface cards (NIC) will communicate with the “nearest” AP. Nearest really means the AP with the greatest signal strength at the point where the user is located, which might not physically be the nearest AP. APs form a LAN segment. Wireless stations vie for use of the wireless media in the same way wired stations vie for use of the wired media. The method specified in 802.11b is called carrier sense multiple access with collision avoidance or CSMA/CA. It is a method similar to CSMA/CD first specified in IEEE 802.3, the 
bus standard for wired LANs. 

Voice Digitization Standard: DS0 by Hoang lihuo



  • - Quantization: 256 levels
  • - Sampling: 8,000 samples/second
  • - Coding: 8 bits/sample
  • - “Pulse Code Modulation” (PCM)
  • - 8,000 bytes per second
  • - 64,000 bits/second = 64 kb/s
  • - DS0 rate
    There are three steps in voice digitization: quantization, sampling and coding.
The telephone system quantizes the voice signal to 256 levels. This number is chosen to reduce the quantization error, which would be heard as noise after the signal is reconstructed, so that a person can’t hear it on the line. The diagram shows bin numbers 127 and 128 around zero volts.
The second step is sampling. Since this is a voiceband signal, the frequency bandwidth is about 3000 Hz, and so the sampling rate must be at least 6001 times per second, following the Dr. Nyquist’s sampling theorem. To ensure that there are no aliasing errors, the telephone system samples more often: 8,000 samples per second.
    The third step is coding. The telephone system uses 8 bits to code the value of each sample. This technique of using 8 bits per sample is called by some Pulse Code Modulation (PCM), which doesn’t really mean anything. To determine the number of bits per second required, multiply the number of samples per second (8,000) by the number of bits per sample (8) to get 64,000 bits per second, or 64 kb/s for short. This 64 kb/s rate is called a DS0 rate signal (Digital service level zero, or digital signal rate zero, just called “DS0s” in the business). This is the base rate of most transmission systems and digital voice systems. When someone talks about a channel on a digital transmission system, they usually mean a DS0.
Commercial transmission systems which are actually deployed were designed to carry digitized voice, and thus move multiple DS0s. Since they are digital systems, they can be easily be adapted to carry data or video as well as digitized voice.
    The bottom line: we move a byte (representing the value of the sample) 8,000 times/second from one end to the other, for each voice.