Passive VoIP Call Recording

Passive VoIP Call Recording

UNDERSTANDING THE TECHNICAL CHALLENGES OF VOIP RECORDING

DECEMBER 6, 2004

Karen Van BlarcumSr. Technical Writer

VOIP CALL RECORDING

© 2006 AudioCodes USA, Inc · All Rights Reserved

AbstractSince the mid-1990s, Voice Over IP (VoIP) has steadily changed the telecommunications industry. The convergence of data and voice in the communications market allows for value-added ser-vices not available on traditional circuit-based networks, not to mention cost saving advantages. VoIP technology enables busi-nesses to reduce costs, consolidate and simplify networks, and improve customer service applications. VoIP, once viewed as just a new technology, is now recognized as a reliable and cost-effective business solution.

What does it mean to the call recording industry? To remain com-petitive, businesses that develop call recording applications must now implement VoIP solutions. This paper introduces VoIP record-ing and differentiates it from traditional circuit-based recording. It begins with an overview of the IP telephony network then exam-ines the unique challenges of VoIP call recording. Finally, the paper presents AudioCodes’ VoIP recorders, a suite of hardware compo-nents designed to support VoIP call recording applications.

2

VoIP Call Recording403-0028-001. 01 Build 041206 Rev. Bwww.audiocodes.com/bladesAudio Codes verifies the accuracy of documentation when published. Information is subject to change without notice.

http://www.musictelecom.com

VOIP CALL RECORDING


An Overview of VoIPVoIP - also known as Internet telephony, IP telephony, packet-voice, packetized voice, or voice over IP - transmits voice traffic in the form of packets. Since VoIP is reliable and efficient, call centers seeking to improve customer service and to reduce network costs have adopted it. Looking ahead, call recording businesses are expected to do the same.

Hierarchical VoIP Network Structure

A typical IP network consists of interconnected routers that form a packet switching fabric. VoIP is designed to take advantage of this IP infrastructure. There are many ways to add VoIP tech-nology to a LAN network. The simplest design requires the addition of a VoIP call control server, such as a Call Agent. This server provides the logic and control func-tions required to maintain the call state. In this scenario, the phone call from the Internet enters the local net-work via the router. Signaling information passes to the Call Agent, which then sets up and manages the call. The voice conversation passes directly from the router to the IP phone via LAN switches. Unlike circuit-based systems, where voice traffic passes along the same cable as signaling traffic, VoIP technology separates the two.

Hybrid Networks

VoIP networks can also be designed to interface with a conventional PSTN network, usually a T1 or E1 line. In this situation, a Gateway is used to convert traffic between the two networks. In some scenarios, the local phone network consists entirely of IP telephones and a Call Agent manages call states. In other environments, the local phone system is a combination of VoIP and conven-tional PSTN phones. In this case, call control requires both a Call Agent, and a conventional PBX. Alternatively, many manufacturers are designing hybrid PBXs so that VoIP and PSTN phones can coexist.

Call Agent

Voice Packets

Call Control

3



VOIP CALL RECORDING


Integrating Distant Offices

VoIP technology enables businesses with distant offices to reduce operating costs by consolidating and simplifying network design. Many companies, specifi-cally those with world-wide call centers, are adopting VoIP technology for this very reason. As a hypothetical example, take a call center that has three offices (segments) located in California, New York and Texas. With VoIP technology, a single Call Agent manages call control on all three networks while the local net-work’s existing Ethernet switches voice traffic to/from IP phones. Operational costs decrease dramatically because a separate telephone network is no longer required. The following diagram illustrates the efficiency of VoIP technology:

What do Customers Expect from VoIP Recording?

For a business that purchases a call recorder, VoIP simply allows networks to carry telephone conversations. Customers who already have a conventional recording system expect enhanced capabilities from a new VoIP recorder. Cus-tomers who are new to call recording have their unique business requirements in mind and are looking to solve their business objectives. Ultimately, customers focus on the recording product’s features, rather than on the underlying VoIP technology.

Application developers who design call recording systems recognize VoIP technol-ogy’s ascendancy. It is a fact: to remain competitive in the call recording market, a VoIP recorder is a must. This recorder must be able to provide at least the same features available on PSTN recording applica-tions.

Internet

Router C

4



VOIP CALL RECORDING


All call recording systems rely on hardware components that tap into the tele-phone network and pull data into a recording application. Similarly, all recording applications require the same data: the voice conversation (for recording pur-poses), the call control information (to monitor call states), and data for value added services (DTMF, CallerID, etc.). So how do call recorders designed for a VoIP network differ from other call recording systems?

What’s Different about a VoIP Network?

VoIP’s packet-based network presents a new tapping environment with a unique set of challenges. When designing a VoIP recording system, it is important to carefully research these differences and plan for them. This section has been written for call recording companies: whether they are new to call recording; migrating a PSTN recorder to the VoIP network; or improving the performance of an early stage VoIP recording system.

Jitter & Synchronization

One of the most significant differences introduced by VoIP is how audio data arrives. On a conventional circuit-based network, once a call is established, the physical path between the two end points is fixed. (On analog systems both up-stream and down-stream traffic are carried on the same wire and are presented as waveform. On digital systems, up-stream and down-stream traffic are carried on separate wires, but are synchronized to prevent interruptions within the call). In the IP world, the two end points are not fixed and are viewed as connectionless. Media RTP packets carrying voice data for a single call can be routed through dif-ferent paths. As a result, packets of voice data arrive at the end point at different times (jitter) and out of sequence.

To compensate for jitter, IP data networks use buffers to store incoming packets. This gives delayed packets time to ‘catch up’ before the data is eventually sorted and passed to the end user. This system is designed for networks where real time guarantees are not required and delays in packet delivery is acceptable. However, on a telephone network delayed packets reduce voice quality. Packet buffering, though required on a VoIP network, must meet or exceed the standards of a tele-phone network (a maximum delay of 500 ms).

Consider the following scenario: An ethernet cable is tapped for voice packets and the VoIP recorder intercepts the packets before they have been buffered. As a result, the packets pulled off the network are mis-aligned and predictably, the audio quality is poor. To compensate for this, hardware components used for VoIP call recording properly time the buffering of incoming packets.

Packet Filtering

In a conventional circuit-based telephone network, the line is used to transmit only voice data. On an IP network many types of packets - data, voice and media - are present on the same ethernet cable. Packet filtering is the selective passing or blocking of packets as they pass through a network interface. Packet filtering is used by VoIP recording systems to isolate voice related packets from data and media packets.

5



VOIP CALL RECORDING


Many early-stage VoIP recorders rely on host resources for packet filtering. This is a viable solution on networks with light traffic. However, this system is not scal-able and quickly reaches limitation when the system density grows beyond 100 ports. The better solution is a logging system that uses hardware components capable of packet filtering. This system would no longer be limited by host resources and would provide a scalable solution for either low to high density environments.

Voice CODECS

An important consideration in the design any logging system is its ability to encode/decode numerous compression schemes. Like all recording environments, the type of CODEC used for media transport is controlled by the network. As a result, when selecting hardware components for call recording, application devel-opers prefer products that support multiple CODECs. This is crucial when tapping a VoIP network. When call setup is negotiated between two Call Agents, the media format is also negotiated. As a result, the type of media format used can change from call to call on one network. Unlike circuit-based recording systems, a VoIP recorder must have the ability to determine the type of media format on a per call basis. This is accomplished by decoding the packet’s header, where the media format is identified. In today’s VoIP market, the formats G.711, G.723.1, or G.729A are prevalent on most VoIP networks and must be supported by the recording hardware.

The type of media format used for recording is driven by the business needs of the customer. As an application developer, you may be asked to design one sys-tem that maximizes storage capabilities and then another system that requires web-enabled playback. The best bet is a versatile hardware component capable of encoding a variety of media formats. Specifically, components that offer both low bit rate CODECS and .wav header support.

Signaling

All call recording applications rely on hardware components to interpret call con-trol and signaling information. Generally speaking, most applications monitor call states to observe line activity and control the recording process. Other applica-tions are designed to monitor the caller’s experience or agent behavior. These recorders rely on detailed information, such as hold states, to complete their task.

Tapping into a VoIP network requires a component capable of decoding VoIP pro-tocols. More than one type of protocol is used on VoIP networks but the most common are H.323 and SIP. Also, many PBX manufacturers are designing propri-etary protocols to manage call control between the PBX and IP phones. Cisco’s SCCP (Skinny) is one example. The best solution would be to design a call logging system around a hardware component capable of decoding standard and propri-etary VoIP environments. When designed properly, this single solution can inte-grate with any VoIP network.

6



VOIP CALL RECORDING


Transporting DTMF

A DTMF (Dual Tone Multiple Frequency) signaling system detects touch-tone dial-ing. When a button on a touch-tone phone is pressed, the tone is generated, com-pressed, transported to the other party, and then decompressed. On VoIP networks, which use low-bandwidth CODECs, the tone may be distorted during compression and decompression. To address this, VoIP protocols now include a relay method that allows for out-of-band DTMF delivery. Relay methods vary from network and include the following:

•Real-Time Transport Protocol (RTP) can be used to carry specially marked RTP packets. Here the DTMF tones are sent in the same RTP channel as the voice data. The DTMF tones are encoded differently from the voice samples and are identified by a different RTP payload type code.

•When H.323 is used, either the H.245 signal or H.245 alphanumeric method is available. These methods separate DTMF digits from the RTP channel and send them through the H.245 signaling channel.

•Using Named Telephone Events (NTE). Using NTE to relay DTMF tones provides a standardized means of transporting DTMF tones as RTP pack-ets. With the NTE method, the endpoints perform per-call negotiation of the DTMF relay method.

When a VoIP network is deployed, the user can select preferred DTMF delivery methods. Please keep in mind, however, that calls are not processed uniformly. There are cases when the actual delivery method differs from the preferred deliv-ery method. This underscores the importance of selecting a versatile recording component.

Encryption

Companies that have experienced security problems with their data networks are concerned about security with VoIP. There are standards for encrypting data on VoIP networks and some companies are using them. What does this mean to the call recording industry? That depends on the type of encryption method deployed.

Companies typically encrypt data passing between office locations over a VPN. The data encryption/decryption takes place at the endpoints of the VPN - outside of the local network. The data passing along the local network is unsecured. The voice related packets between the VPN and the IP phones are not encrypted. A tap positioned anywhere on the local network is capable of recording.

Alternatively, the data could be encrypted at the endpoints - the IP phones. VoIP traffic traveling along the local network is encrypted and cannot be tapped. Today most IP phones lack the processing resources for this type of implementation. It is also expensive for a company to deploy. It is unlikely that a call recording com-pany would encounter this type of environment.

7



VOIP CALL RECORDING


Data Flow

On traditional telephone networks, all voice and call control information passes through a central location - the PBX. Each channel on the network is tapped indi-vidually, and a central tapping system obtains all voice and call control informa-tion on the local network. With VoIP, when an incoming or outgoing call is initiated, only the call control information is passed along the ethernet to the Call Agent. After call setup is complete, the voice packets are passed to the end point - either a phone on the external network or a local IP phone. An IP network does not have a central location where all voice and call control information converges. The following diagrams illustrate this point.

Router or Gateway

External Networks

An incoming call enters the external facing Router or Gateway. The call control passes to the Call Agent, which then negotiates the call with the local IP phone. Once the call is connected, the voice packets pass directly to the phone.

Voice Packets

Call Control

Router or Gateway

External Networks

Agent 1 initiates a call to Agent 2. Call control information passes to the Call Agent. Once the call is initiated, the voice packets pass directly to the other local IP phone. The two phones are connected to the same switch, so the voice packets do not leave this LAN segment.

Voice Packets

Call Control

t 1 Agent 2

8



VOIP CALL RECORDING


In general, the type of information required by the call recording application determines the location of the tap. Many call recorders record all calls entering or leaving the local telephone network. Here, the tap point is located between the Central Office (CO) and the local PBX. This is commonly referred to as “trunk recording”. Other call recording applications monitor agent behavior. Here the tap is located between the local PBX and agent phones, so that local call control information passes into the recording application. The following diagrams illus-trate these two tapping scenarios on a VoIP network:

External Networks

Router or Gateway

Trunk Recording: A tap point is positioned internally on the network directly behind the outside facing router or Gateway. All voice traffic entering or leaving the local phone network is recorded through this point. Call control information passing between the external network and the Call Agent is captured. Internal calls (agent to agent calls) and call control passing from the phones to the Call Agent is not captured.

TAP

ernal works

uter or Gateway

Monitoring Agent Behavior: One tap is placed between the Call Agent and switch leading to IP phones. In this scenario, all voice traffic leaving and entering the local network is recorded, as well as all call control information. Agent behavior is monitored through call control information passing from the IP phones to the Call Agent. Voice packets passing between IP phones are not captured.

TAP

TAP

9



VOIP CALL RECORDING


Local (Agent to Agent) Recording

Some call monitoring applications record all phone conversations - including agent to agent. As demonstrated in the above illustrations, this type of recording becomes more complicated in a VoIP environment. When a call is placed to another phone on the local network, only the call control information passes to the Call Agent. The voice packets are passed directly between the two IP phones. If the two phones are connected to the same switch, voice packets never leave that segment of the network.

If local recording is required, the tap points must be distributed throughout the network. One option would be to install taps on each individual phone on the net-work. Though 100% effective, this is expensive. A second option is to tap the span (mirror) port of each switch. Here, a recording application captures both call control and voice packets for each phone. Unfortunately, span ports support data flow at the rate of 100 mbs. Data is passing through the ethernet at a rate of 100 mbs in both directions. This tap point reaches a bandwidth limit when the net-work operates at 50% capacity.

To address this limitation use the span port so that information only passes in one direction. Then rely on a high impedance tap installed on the ethernet cable to pull data transmitted from the other direction. In this sce-nario, the recording application retrieves 100% of call control and voice packets for each IP phone con-nected to the switch.

Distributing a VoIP Call Recording System

The introduction of VoIP dramatically changes telephony architecture. Where con-ventional PSTN networks are deployed with a standard architecture, IP based telephone networks are not. There are endless ways to design a corporate net-work, and now the same can be said for telephone networks. Call recording com-panies are forced to look beyond a central tapping solution and work with a flexible approach. If designed well, a single VoIP call recording system can be reused on another network with minimal development effort. Call recorders cre-ated with a modular design are the most flexible and provide the best long term approach when planning a VoIP recording solution.

External Networks

Router or Gateway

Call Recorder

10



VOIP CALL RECORDING


The following diagrams present two types of distributed VoIP networks. In the first example, tap points and packet filtering resources are distributed on the net-work. The filtered data is then passed via an internal IP network to a centralized recording server. The second illustration shows a call recording system that is dis-tributed throughout the VoIP network. All resources - tap point, decoding, packet filtering and recording - are centralized at each site. Since the architecture of a VoIP network varies dramatically from location to location, the best overall solu-tion is to design a modular call recording system.

Recording Resources

Tap points are distributed throughout the local VoIP network and connected to packet filtering resources. From here, all voice related packets are passed on an internal network to a centralized location for recording.

Packet Filtering

Packet Filtering

Packet Filtering

Internet

Call RecorderCall Recorder Call RecorderCall Recorder Call RecorderCall Recorder

A large corporation has three office segments controlled by a single Call Agent. Here taps are distributed throughout the three office segments and provide local packet filtering, decoding and recording resources.

11



VOIP CALL RECORDING


Summary

VoIP’s packet-based network presents a new tapping environment with a unique set of challenges. This paper has introduced call recording companies to many of the technical aspects associated with VoIP recording. Whether your company is new to call recording, migrating an existing PSTN recorder to a VoIP network, or improving an existing VoIP recorder, it is important to carefully research these dif-ferences and build them into the design of your VoIP recording system.

AudioCodes is a proven market leader in call recording technology. The Smart-WORKS product line has set world-wide standard for circuit based tapping com-ponents. With the introduction of VoIP recorders, AudioCodes USA is extending it’s product line to the IP environment. Like all of AudioCodes USA passive tap-ping components, these products are targeted for high performance recording in low to high density environments.

AudioCodes developed the SmartWORKS IPX as a tool that integrates with the underlying VoIP technology and passes information to a call recording application. Like all passive taps, the IPX is capable of capturing call sessions on the network, decoding call control or signaling information, plus provides a mechanism for encoding/decoding voice data. These capabilities, along with the concepts explained in this document, have been carefully addressed and incorporated into the design of the SmartWORKS IPX.

12



VOIP CALL RECORDING


About AudioCodes USAAudioCodes USA Inc., is a leading provider of advanced voice recording technology and CT integra-tion cards for the voice/data logging and call recording markets.

Our products are widely used by many of the major call recording and voice/data logging companies around the globe. At AudioCodes, we have a ‘customer comes first’ philosophy, and often encourage feedback from our customers to help us assess their needs. We are recognized for our commitment to building long lasting relationships, providing quality service, creating unique solutions, and maintaining our customers’ satisfaction.

AudioCodes USA, Inc. is owned by AudioCodes, Ltd. (Nasdaq: AUDC), a leading provider of innova-tive, reliable and cost-effective Voice over Packet technology and Voice Network products.

Headquarters 27 World’s Fair DriveSomerset, NJ 08873732.469.0880www.audiocodes.com/blades

For more information about VoIp Recording contact:

Pin Lo Chen: [email protected]

Disclaimer© 2004 AudioCodes USA, Inc. All rights reserved. AudioCodes, the AudioCodes logo and SMART-WORKS are trademarks or registered trademarks of AudioCodes Inc. All other marks are the prop-erty of their respective owners. The information and specifications in this document and the product(s) are subject to change without notice.

13



Documents

Passive VoIP Call Recording