VoIP Fundamentals (Introducing Voice over IP Networks) Part 1

Voice over Internet Protocol (VoIP) allows a voice-enabled router to carry voice traffic, such as telephone calls and faxes, over an Internet Protocol (IP) network. This topic introduces the fundamentals of VoIP, the various types of voice gateways, and how to use gateways in different IP telephony environments.

Voice over IP is also known as VoIP. You might also hear VoIP referred to as IP Telephony. Both terms refer to sending voice across an IP network. However, the primary distinction revolves around the endpoints in use. For example, in a VoIP network, traditional analog or digital circuits connect into an IP network, typically through some sort of gateway. However, an IP telephony environment contains endpoints that natively communicate using IP. Be aware that much of the literature on the subject, including this topic, might use these terms interchangeably.

VoIP routes voice conversations over IP-based networks, including the Internet. VoIP has made it possible for businesses to realize cost savings by utilizing their existing IP network to carry voice and data, especially where businesses have underutilized network capacity that can carry VoIP at no additional cost. This section introduces VoIP, the required components in VoIP networks, currently available VoIP signaling protocols, VoIP service issues, and media transmission protocols.

Cisco Unified Communications Architecture

The Cisco Unified Communications System fully integrates communications by enabling data, voice, and video to be transmitted over a single network infrastructure using standards-based IP. Leveraging the framework provided by Cisco IP hardware and software products, the Cisco Unified Communications System has the capability to address current and emerging communications needs in the enterprise environment. The Cisco Unified Communications family of products is designed to optimize feature functionality, reduce configuration and maintenance requirements, and provide interoperability with a variety of other applications. The Cisco Unified Communications System provides and maintains a high level of availability, quality of service (QoS), and security for the network.

The Cisco Unified Communications System incorporates and integrates the following communications technologies:

■ IP telephony: IP telephony refers to technology that transmits voice communications over a network using IP standards. Cisco Unified Communications System includes hardware and software products such as call processing agents, IP phones (both wired and wireless), voice messaging systems, video devices, and other special applications.

■ Customer contact center: Cisco IP Contact Center products combine strategy with architecture to enable efficient and effective customer communications across a global network. This allows organizations to draw from a broader range of resources to service customers. These resources include access to a large pool of customer service agents and multiple channels of communication as well as customer self-help tools.

■ Video telephony: The Cisco Unified Video Advantage products enable real-time video communications and collaboration using the same IP network and call processing agent as Cisco Unified Communications. With Cisco Unified Video Advantage, making a video call is just as easy as dialing a phone number.

■ Rich-media conferencing: Cisco Conference Connection and Cisco Unified MeetingPlace enhance the virtual meeting environment with an integrated set of IP-based tools for voice, video, and web conferencing.

■ Third-party applications: Cisco works with other companies to provide a selection of third-party IP communications applications and products. This helps businesses focus on critical needs such as messaging, customer care, and workforce optimization.

VoIP Overview

VoIP is the family of technologies that allows IP networks to be used for voice applications, such as telephony, voice instant messaging, and teleconferencing. VoIP defines a way to carry voice calls over an IP network, including the digitization and packetization of the voice streams. IP Telephony VoIP standards create a telephony system where higher-level features such as advanced call routing, voice mail, and contact centers can be utilized.

VoIP services convert your voice into a digital signal that travels over an IP-based network. If you are calling a traditional phone number, the signal is converted to a traditional telephone signal before it reaches its destination. VoIP allows you to make a call directly from a computer, a VoIP phone, or a traditional analog phone connected to a special adapter. In addition, wireless "hot spots" in locations such as airports, parks, and cafes that allow you to connect to the Internet might enable you to use VoIP services.

Business Case for VoIP

The business advantages that drive the implementation of VoIP networks have changed over time. Starting with simple media convergence, these advantages evolved to include call-switching intelligence and the total user experience.

Originally, ROI calculations centered on toll-bypass and converged-network savings. Although these savings are still relevant today, advances in voice technologies allow organizations and service providers to differentiate their product offerings by providing the following:

■ Cost savings: Traditional time-division multiplexing (TDM), which is used in the public switched telephone network (PSTN) environment, dedicates 64 kbps of bandwidth per voice channel. This approach results in bandwidth being unused when no voice traffic exists. VoIP shares bandwidth across multiple logical connections, which results in a more efficient use of the bandwidth, thereby reducing bandwidth requirements. A substantial amount of equipment is needed to combine 64-kbps channels into high-speed links for transport across a network. Packet telephony uses statistical analysis to multiplex voice traffic alongside data traffic. This consolidation results in substantial savings on capital equipment and operations costs.

■ Flexibility: The sophisticated functionality of IP networks allows organizations to be flexible in the types of applications and services they provide to their customers and users. Service providers can easily segment customers. This helps them to provide different applications, custom services, and rates depending on traffic volume needs and other customer-specific factors.

■ Advanced features: Following are some examples of the advanced features provided by current VoIP applications:

■ Advanced call routing: When multiple paths exist to connect a call to its destination, some of these paths might be preferred over others based on cost, distance, quality, partner handoffs, traffic load, or various other considerations. Least-cost routing and time-of-day routing are two examples of advanced call routing that can be implemented to determine the best possible route for each call.

■ Unified messaging: Unified messaging improves communications and productivity. It provides a single user interface for messages that have been delivered over a variety of mediums. For example, users can read their e-mail, hear their voice mail, and view fax messages by accessing a single inbox.

■ Integrated information systems: Organizations use VoIP to affect business process transformation. These processes include centralized call control, geographically dispersed virtual contact centers, and access to resources and self-help tools.

■ Long-distance toll bypass: Long-distance toll bypass is an attractive solution for organizations that place a significant number of calls between sites that are charged traditional long-distance fees. In this case, it might be more cost-effective to use VoIP to place those calls across an IP network. If the IP WAN becomes congested, calls can overflow into the PSTN, ensuring that no degradation occurs in voice quality.

■ Security: Mechanisms in an IP network allow an administrator to ensure that IP conversations are secure. Encryption of sensitive signaling header fields and message bodies protect packets in case of unauthorized packet interception.

■ Customer relationships: The capability to provide customer support through multiple mediums, such as telephone, chat, and e-mail, builds solid customer satisfaction and loyalty. A pervasive IP network allows organizations to provide contact center agents with consolidated and up-to-date customer records along with related customer communication. Access to this information allows quick problem solving, which builds strong customer relationships.

■ Telephony application services: XML services on Cisco IP Phones give users another way to perform or access business applications. Some examples of XML-based services on Cisco IP Phones are user stock quotes, inventory checks, direct-dial directory, announcements, and advertisements. Some Cisco IP Phones are equipped with a pixel-based display that can display full graphics instead of just text in the window. The pixel-based display capabilities allow you to use sophisticated graphical presentations for applications on Cisco IP Phones and make them available at any desktop, counter, or location.

Components of a VoIP Network

Figure 1-1 depicts the basic components of a packet voice network.

Components of a VoIP Network

Figure 1-1 Components of a VoIP Network

The following is a description of these basic components:

■ IP Phones: Cisco IP Phones provide IP endpoints for voice communication.

■ Gatekeeper: A gatekeeper provides Call Admission Control (CAC), bandwidth control and management, and address translation.

■ Gateway: The gateway provides translation between VoIP and non-VoIP networks, such as the PSTN. Gateways also provide physical access for local analog and digital voice devices, such as telephones, fax machines, key sets, and private branch exchanges (PBX).

■ Multipoint Control Unit (MCU): An MCU provides real-time connectivity for participants in multiple locations to attend the same videoconference or meeting.

■ Call agent: A call agent provides call control for IP phones, CAC, bandwidth control and management, and address translation. Unlike a gatekeeper, which in a Cisco environment typically runs on a router, a call agent typically runs on a server platform. Cisco Unified Communications Manager is an example of a call agent.

■ Application servers: Application servers provide services such as voice mail, unified messaging, and Cisco Communications Manager Attendant Console.

■ Videoconference station: A videoconference station provides access for end-user participation in videoconferencing. The videoconference station contains a video capture device for video input and a microphone for audio input. A user can view video streams and hear audio that originates at a remote user station.

Other components, such as software voice applications, interactive voice response (IVR) systems, and soft phones, provide additional services to meet the needs of an enterprise site.

VoIP Functions

In the traditional PSTN telephony network, all the elements required to complete a call are transparent to an end user. Migration to VoIP requires an awareness of these required elements and a thorough understanding of the protocols and components that provide the same functionality in an IP network.

Required VoIP functionality includes these functions:

■ Signaling: Signaling is the capability to generate and exchange control information that will be used to establish, monitor, and release connections between two end-points. Voice signaling requires the capability to provide supervisory, address, and alerting functionality between nodes. The PSTN network uses Signaling System 7 (SS7) to transport control messages. SS7 uses out-of-band signaling, which, in this case, is the exchange of call control information in a separate dedicated channel.

VoIP presents several options for signaling, including H.323, Session Initiation Protocol (SIP), H.248, Media Gateway Control Protocol (MGCP), and Skinny Client Control Protocol (SCCP). Some VoIP gateways are also capable of initiating SS7 signaling directly to the PSTN network. Signaling protocols are classified as either peer-to-peer or client/server protocols.

SIP and H.323 are examples of peer-to-peer signaling protocols where the end devices or gateways contain the intelligence to initiate and terminate calls and interpret call control messages. H.248, SCCP, and MGCP are examples of client/server protocols where the endpoints or gateways do not contain call control intelligence but send or receive event notifications to a server commonly referred to as a call agent. For example, when an MGCP gateway detects a telephone that has gone off hook, it does not know to automatically provide a dial tone. The gateway sends an event notification to the call agent, telling the agent that an off-hook condition has been detected. The call agent notifies the gateway to provide a dial tone.

■ Database services: Access to services, such as toll-free numbers or caller ID, requires the capability to query a database to determine whether the call can be placed or information can be made available. Database services include access to billing information, caller name delivery (CNAM), toll-free database services, and calling-card services. VoIP service providers can differentiate their services by providing access to many unique database services. For example, to simplify fax access to mobile users, a provider can build a service that converts fax to e-mail. Another example is providing a call notification service that places outbound calls with prerecorded messages at specific times to notify users of such events as school closures, wake-up calls, or appointments.

■ Bearer control: Bearer channels are the channels that carry voice calls. Proper supervision of these channels requires that appropriate call connect and call disconnect signaling be passed between end devices. Correct signaling ensures that the channel is allocated to the current voice call and that a channel is properly deallocated when either side terminates the call. Connect and disconnect messages are carried by SS7 in the PSTN network. Connect and disconnect message are carried by SIP, H.323, H.248, or MGCP within the IP network.

■ Codecs: Codecs provide the coding and decoding translation between analog and digital facilities. Each codec type defines the method of voice coding and the compression mechanism that is used to convert the voice stream. The PSTN uses TDM to carry each voice call. Each voice channel reserves 64 kbps of bandwidth and uses the G.711 codec to convert an analog voice wave to a 64-kbps digitized voice stream. In VoIP design, codecs might compress voice beyond the 64-kbps voice stream to allow more efficient use of network resources. The most widely used codec in the WAN environment is G.729, which compresses the voice stream to 8 kbps.

VoIP Signaling Protocols

VoIP uses several control and call-signaling protocols. Among these are:

■ H.323: H.323 is a standard that specifies the components, protocols, and procedures that provide multimedia communication services, real-time audio, video, and data communications over packet networks, including IP networks. H.323 is part of a family of International Telecommunication Union Telecommunication Standardization sector (ITU-T) recommendations called H.32x that provides multimedia communication services over a variety of networks. H.32x is an umbrella of standards that define all aspects of synchronized voice, video, and data transmission. It also defines end-to-end call signaling.

■ MGCP: MGCP is a method for PSTN gateway control or thin device control. Specified in RFC 2705, MGCP defines a protocol that controls VoIP gateways that are connected to external call control devices, referred to as call agents. MGCP provides the signaling capability for less-expensive edge devices, such as gateways, that might not have implemented a full voice-signaling protocol such as H.323. For example, anytime an event, such as off-hook, occurs on a voice port of a gateway, the voice port reports that event to the call agent. The call agent then signals the voice port to provide a service, such as dial-tone signaling.

■ SIP: SIP is a detailed protocol that specifies the commands and responses to set up and tear down calls. SIP also details features such as security, proxy, and transport control protocol (TCP) or User Datagram Protocol (UDP) services. SIP and its partner protocols, Session Announcement Protocol (SAP) and Session Description Protocol (SDP), provide announcements and information about multicast sessions to users on a network. SIP defines end-to-end call signaling between devices. SIP is a text-based protocol that borrows many elements of HTTP, using the same transaction request and response model and similar header and response codes. It also adopts a modified form of the URL addressing scheme used within e-mail that is based on Simple Mail Transfer Protocol (SMTP).

■ SCCP: SCCP is a Cisco proprietary protocol used between Cisco Communications Manager and Cisco IP Phones. The end stations (telephones) that use SCCP are called Skinny clients, which consume less processing overhead. The client communicates with the Cisco Unified Communications Manager (often referred to as Call Manager, abbreviated UCM) using connection-oriented (TCP-based) communication to establish a call with another H.323-compliant end station.

The H.323 Umbrella

H.323 is a suite of protocols defined by the International Telecommunication Union (ITU) for multimedia conferences over LANs. The H.323 protocol was designed by the ITU-T and was initially approved in February 1996. It was developed as a protocol that provides IP networks with traditional telephony functionality. Today, H.323 is the most widely deployed standards-based voice and videoconferencing standard for packet-switched networks.

The protocols specified by H.323 include the following:

■ H.225 Call Signaling: H.225 call signaling is used to establish a connection between two H.323 endpoints. This is achieved by exchanging H.225 protocol messages on the call-signaling channel. The call-signaling channel is opened between two H.323 endpoints or between an endpoint and an H.323 gatekeeper.

■ H.225 Registration, Admission, and Status: Registration, admission, and status (RAS) is the protocol between endpoints (terminals and gateways) and gatekeepers. RAS is used to perform registration, admission control, bandwidth changes, status, and disengage procedures between endpoints and gatekeepers. A RAS channel is used to exchange RAS messages. This signaling channel is opened between an end-point and a gatekeeper prior to the establishment of any other channels.

■ H.245 Control Signaling: H.245 control signaling is used to exchange end-to-end control messages governing the operation of an H.323 endpoint. These control messages carry information related to the following:

■ Capabilities exchange

■ Opening and closing of logical channels used to carry media streams

■ Flow-control messages

■ General commands and indications

■ Audio codecs: An audio codec encodes the audio signal from a microphone for transmission by the transmitting H.323 terminal and decodes the received audio code that is sent to the speaker on the receiving H.323 terminal. Because audio is the minimum service provided by the H.323 standard, all H.323 terminals must have at least one audio codec supported, as specified in the ITU-T G.711 recommendation (coding audio at 64 kbps). Additional audio codec recommendations such as G.722 (64, 56, and 48 kbps), G.723.1 (5.3 and 6.3 kbps), G.728 (16 kbps), and G.729 (8 kbps) might also be supported.

■ Video codecs: A video codec encodes video from a camera for transmission by the transmitting H.323 terminal and decodes the received video code on a video display of the receiving H.323 terminal. Because H.323 specifies support of video as optional, the support of video codecs is optional as well. However, any H.323 terminal providing video communications must support video encoding and decoding as specified in the ITU-T H.261 recommendation.

In Cisco IP Communications environments, H.323 is widely used with gateways, gatekeepers, and third-party H.323 clients, such as video terminals. Connections are configured between devices using static destination IP addresses.

Note Because H.323 is a peer-to-peer protocol, H.323 gateways are not registered with Cisco Unified Communications Manager as an endpoint is. An IP address is configured in the Cisco UCM to confirm that communication is possible.

MGCP is a client/server call control protocol built on a centralized control architecture. MGCP offers the advantage of centralized gateway administration and provides for largely scalable IP telephony solutions. All dial plan information resides on a separate call agent. The call agent, which controls the ports on the gateway, performs call control. An MGCP gateway does media translation between the PSTN and VoIP networks for external calls. In a Cisco-based network, Communications Managers function as call agents.

MGCP is a plain-text protocol used by call-control devices to manage IP telephony gateways. MGCP was defined under RFC 2705, which was updated by RFC 3660, and superseded by RFC 3435, which was updated by RFC 3661.

With MGCP, Cisco UCM knows of and controls individual voice ports on an MGCP gateway. This approach allows complete control of a dial plan from Cisco UCM and gives Communications Manager per-port control of connections to the PSTN, legacy PBX, voice-mail systems, and POTS phones. MGCP is implemented with use of a series of plain-text commands sent via User Datagram Protocol (UDP) port 2427 between the Cisco UCM and a gateway.

It is important to note that for an MGCP interaction to take place with Cisco UCM, an MGCP gateway must have Cisco UCM support. If you are a registered customer of the Software Advisor, you can use this tool to make sure your platform and your Cisco IOS software or Cisco Catalyst operating system version are compatible with Cisco UCM for MGCP. Also, make sure your version of Cisco UCM supports the gateway.

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