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Internet Protocol Centrex
Interactive Communications over Internet Protocol
The Internet Protocol (IP) was devised in the 1960s and has been subse-quently revised as a universal networking language for the network of net-works that we now know as the Internet.Some of the relative objectives of IP are that it should enable:
◆ Data to be transmitted over different media and data links; ◆ Data packets to be transmitted reliably;
◆ New networks to be joined to the Internet without disruption; ◆ Equipment of different types and from various vendors to be
inter-connected smoothly. Layer 3 and 4 Protocols
IP exists at layer 3 (the network layer) of the seven-layered Open System Interconnection (OSI) protocol stack, above the data link layer, where technologies such as ATM, frame relay, and Ethernet apply, as illustrated in Figure 2.1.
While IP, at layer 3, defines the addressing and routing rules for data packets, at layer 4 there are three other relevant protocols: user datagram protocol (UDP), Real-Time Transport Protocol (RTP), and transmission control protocol (TCP).UDP is a connectionless protocol, with an emphasis on the minimizing of packet transit time, which is ideal for the interactions that are essential to voice and video traffic.TCP is the connection-oriented protocol that provides confirmation of a packets receipt at its destination, if necessary after retransmission and without con-cern about delay.With a Voice or Video-Conferencing over Internet Pro-tocol (VoIP) session it is usual to employ TCP/IP for the call setup, any in-progress signaling, and teardown, while UDP/IP is better suited to the real-time transmission of the voice or video conversations.TCP provides the same functionality on networks for VoIP as signaling system number 7 (SS7) does over circuit-switched, TDM networks for conventional, long-distance telephony.
RTP is an Internet Engineering Task Force (IETF) standard that defines the streaming of interactive voice and video packets over switched networks.This is a so-called thin protocol that does not provide any quality of service (QoS) features.
IP Telephony
For several years the term Voice over Internet Protocol (VoIP) has been used to identify the transmission of voice signals over any IP-based
Layer 7 6 5 4 3 2 1 Layer 4 3 2 1 Network interface: Ethernet, ATM, FR Internet: IP Transport: RTP, TCP, UDP Application: Web Browser, HTTP IP OSI Physical Data link Network Transport Session Presentation Application
network.In the business context, VoIP now refers to the transmission of voice and/or video communications over a managed WAN or LAN.
Some consultants and one or two telecom manufacturers have used the term telephony over Internet protocol (ToIP) to describe the switching of real-time, conversational traffic in systems that are attached to IP data networks.
The words IP telephony have emerged as the umbrella term that cov-ers both VoIP and ToIP, which can be delivered either by an IP-PBX (housed on the customers premises) or by IP-Centrex (for which the call processor is owned and accommodated by the carrier).
Some organizations have been using VoIP to reduce the cost of long-distance service, particularly international service, for several years.In countries that have highly competitive interexchange carriers it is generally no longer worthwhile to use VoIP, after allowing for the cost of gate-way hardware and software.Several international carriers, such as the T-Systems division of Deutsche Telekom, have specifically promoted the use of IP networks for international traffic.
IP telephony has been validated since the year 2000 by the availability of some IP-PBXs (see Chapter 4).An IP-PBX system or IP-Centrex service may deliver only VoIP (through appropriate trunk and line interfaces), only ToIP (by retaining interfaces for the existing analog and digital devices), or both in the form of IP telephony.Now that IP has become, by far, the most popular protocol for data transmission and is widely deployed in networks of all sizes, from one room to worldwide, organizations have access to the appropriate transport technology to gain the advantages of convergence.Full convergence between multimedia and data applications requires the availability of both VoIP and ToIP, which are both inherent to the concept of IP-Centrex.
Interface Standards
The key to success with IP-Centrex will be a high level of interoperability between devices and systems or applications.This will be a major move away from the proprietary interfaces that have kept acquisition costs up and made application implementations complex with PBX systems.The acceptance of a limited number of open standards will also facilitate more competition between a broader range of manufacturers and service providers.
The earliest call control standard for mapping users names or tele-phone numbers into and IP source or destination addresses was H.323,
which was adopted by the International Telecommunications Union (ITU).H.323, was originally intended to define how multimedia commu-nications were to be transmitted over a data network between teleconfer-ence units and, as used for VoIP, defines only a restricted feature set, with different enhancements being made by various manufacturers.H.323 does not define transmitted voice quality, is considered to be too processing intensive, and depends on the use of intelligent workstations.
A working group of the IETF created the session initiation protocol (SIP) to lessen call setup times and take better advantage of the Internet infrastructure than H.323. SIP is most likely to become the interface stan-dard of choice between telephone sets and computers with ToIP systems.
A third standard, officially known as H.248 but more generally as MGCP or Megaco, is being jointly developed by the IETF and ITU, with support from some, but not all, major manufacturers.The H.248/ MGCP protocol addresses the needs of multimedia conferencing and is intended for use with media gateway controllers (MGC).
Every communicating device on an Internet protocol-conforming net-work must have an IP address, so that a desktop with a telephone, a PC, and a softphone (within the PC) or a video terminal needs to be allocated three addresses.There must be a process with the network for mapping tele-phone numbers to corresponding IP addresses.
Also, any endpoints address must be known in order to be accessible; in some circumstances, this requirement becomes a security concern.This situation can be problematic in that multiprotocol label switching (MLPS), which is widely used in managed wide area networks (WANs), does not allow for any spoofing (i.e., the alteration and retransmission of any part of a signal in order to hide the address contents and therefore discourage hacking).For details regarding the use of MPLS, see Managed Networks in Chapter 3.
IP Centrex Configurations
The development of IP-Centrex is quickly following the same path trod by the manufacturers of IP-PBXs over the past few years.There are two main stages in this development, where IP-Centrex is first delivered from an existing, circuit-switched type of serving CO and then, launched from a new controller that handles only packet-switched traffic.
The first stage of IP-Centrex could be described as a hybrid service, supporting existing terminals and new IP phones.We may think of the sec-ond stage as pure IP-Centrex, with a server in the carriers network pro-viding control signals to various gateways and links to application processors (APs), for applications such as unified messaging (UM), auto-matic call distribution (ACD), and CRM.
Hybrid IP-Centrex Service
In the examples that are shown in Figures 2.2 and 2.3, a mixture of conven-tional (analog and digital) and IP phones (both separate sets and soft-phones on PCs) is used in the single or multilocation situations, served by one legacy Centrex system.
At an organizations branch location, as in Figure 2.3, an on-site IP-Centrex gateway may be placed on the customers premises to support
Trunk Toll Serving central office
IP Centrex gateway Broadband access link Class 5 switch Digital Analog ISDN/BRI Laptop Edge router Executive IP phones Simple IP phones Desktop PC (Softphones) Fax machine Customers premises Managed IP network
non-IP phones and, possibly, facsimile machines, through analog and digi-tal ports.
Pure IP-Centrex is based on a new type of CO switch that is now appearing on the market and is frequently known as a softswitch. Part of a typical softswitch configuration is shown in Figure 2.4. With this devel-opment the call control functions are physically separated from the various outboard gateways, in a way that has not been obvious with legacy, circuit-switched CO systems.
Trends to Full IP-Centrex
The softswitch-based IP-Centrex solution must continue to provide for a significant proportion of legacy phones, both analog and digital, while accommodating a growing population of IP-compatible devices.We do not encounter many large greenfield situations, where hundreds of users need new phones or computers at one time, in any one year.Most organi-zations wish to retain their present inventory of phone sets and a few fax machines for several years.A complete switch to local area network (LAN)-based voice and workstations is much more likely in home offices, for telecommuters, and in branch locations, as at the bottom of Figure 2.4. In the ILEC networks the full transition from large CO systems, using
Trunk Toll Serving central office
IP Centrex gateway Managed IP network Broadband links Class 5 switch (Legacy CTX) Digital Analog ISDN/BRI Edge router Executive IP phones Simple IP phones Desktop PC (Softphones) Fax machine Branch offices Head office
TDM technology, to softswitch COs will certainly take a decade and may last up to 20 years.The competitive service providers, with less capital invested in legacy systems, may be able to move to the pure IP-Centrex sce-nario more quickly.
Softswitch Characteristics
A softswitch is a high-availability computing platform that houses software to control multimedia traffic over an integrated telecom network and mediates the signaling between packet and circuit-switched networks (i.e., between the IP-based and the legacy TDM domains).Softswitches may be considered software-based replacements for class 5 (local exchange) and class 4 (toll exchange) central offices.
These systems promise an excellent opportunity for service providers to deliver new, innovative broadband services, while reducing equipment and personnel costs by up to 50%, compared with conventional, digital CO systems.The legacy switches, made by the eight major telecommunication equipment manufacturers, are highly proprietary in nature.
IP phone Laptop Home office CRM ACD UM Application gateway Trunk network gateway Broadband access IP softswitch (CTX controller) IP gateway Managed IP network Broadband accesses PC (soft phone) Branch office IP CTX gateway Digital Analog IP phones IP phones Voice Video Router T3T1
A softswitch should reside in an extremely reliable (i.e., extensively redundant), industry-grade, rack-mounted server in a secure network data center.Ample power supply and network access backup capabilities are also essential.Softswitches work in close conjunction with modular media gateways and, partly because of the physical separation of con-trol and packet-switching functions, can be scaled in capacity to meet quickly changing traffic patterns more easily than with older circuit switches.
A significant feature of softswitches is the facility for a system adminis-trator to install and manage all the telecom services of the switch through a Web portal.In this users are offered a much better way to control their own network, such as ordering new services, click-to-call on any numbers, modifying call forwarding, and customizing individual telephone sets (whether real or virtual).For these reasons, managing telecom services will become much easier than it has been with on-premise PBXs or off-site Centrex services.With a softswitch it should be just as simple to administer video-conference services as voice calls, which will be a big improvement over legacy class 5 CO capabilities.Some of these softswitch entities are illustrated in Figure 2.5.
Carrier grade softswitch products have been on the market since late 2001 from established manufacturers, such as Nortel Networks and Sie-mens, as well as recently started companies including Sonus and Tanqua Systems.Most of the newer companies in this line of business belong to the Softswitch Consortium.
The early applications of softswitches were for off-loading Internet traffic, from more expensive and busy CO systems, primarily with DSL on the line side and IP or ATM on the trunk side.Going forward, the role of softswitches will be much more to replace conventional central office sys-tems and to deliver multimedia IP-Centrex services.Softswitch availability is the key for the development of hosted telephony services by the competi-tive service providing companies and for the capability of putting all an organizations voice, video, and data traffic onto a single network.
Network Considerations for IP-Centrex
Requirements for VoIPThe IP-based systems and networks that will be used for voice or video transmission over IP must meet several criteria to deliver a signal of the
same quality that has been available through circuit-switched, digital sys-tems and with which the users are satisfied.
Because of the real-time, interactive nature of VoIP the average transit time (i.e., latency) between source and destination is most important. As shown in Table 2.1, ideally one-way latency should be less than 100 ms. If the round trip delay on a voice conversation exceeds three-tenths of one second, then it becomes difficult to continue an intelligent dialog, as many have experienced when a telephone call was routed through a
Softswitch IP Centrex customer gateway Broadband access facility Fax machine Trunk
gateway Signalinggateway Analog IP phones Managed IP or ATM network Internet PSTN Figure 2.5 Softswitch configuration.
Table 2.1 VoIP Network Criteria
Criterion Ideal Desirable (max)
Latency 100 ms 150 ms Jitter delay 40 ms 80 ms Packet loss 1% 3%
geostationary satellite.Table 2.1 displays a consensus of numerical criteria published by several manufacturers of VoIP gateway products.
Jitter is the variability of packet arrival times at the receiver and is gen-erally caused by large bursts of data interfering with the real-time traffic in the network.Although modest jitter delays have not been considered seri-ous to real-time conversation, yet significant jitter (i.e. delays approaching 60 ms) leads to unintelligible speech patterns and ruins video displays.
Packet loss corresponds to link dropouts, which is a common experi-ence with mobile/cell phones in congested areas and also results in unsatis-factory voice or video communications.
The bandwidth, or bit rate, requirements for VoIP depend on the type of encoding/decoding (i.e., codec) technique that is used in the digital sig-nal processor (DSP) in the end stations.A reasonably good guideline, with current technology, is to allow 20 Kbps in the network or the access link for each simultaneous voice conversation, or 200 Kbps for a small-group video conference session.
LAN Requirements
In a large office complex there may be a correspondingly large number of simultaneous voice calls on the LAN, in the busy hour of the week.This traffic volume depends on the peak telephone utilization level in the organization, which may be 25% in many offices, but rise to nearly 100% of the population in some highly sales-oriented businesses, such as a stock brokerage.Even if there are 1,000 concurrent voice calls on the LAN, this probably represents an added load of only 20 Mbps, which is insignificant on a gigabit backbone.
However, if the in-building network is to carry IP-based video traffic, as will increasingly be the case as multimedia applications are imple-mented, then a major LAN upgrade is much more likely to be needed.If we assume a minimum bit rate of around 200 Kbps for two-way video interac-tion and up to 2 Mbps for higher definiinterac-tion, full mointerac-tion, video displays, then a number of simultaneous video sessions will put a heavy load on the LAN.In some early installations for IP-based video conferencing a separate LAN that went all the way to the main edge router for the building was dedicated to the video outlets.
WAN Requirements
An existing metropolitan area network (MAN) or WAN will certainly be more challenged by the added burden of carrying VoIP packets than a
LAN.The transit and jitter delays and bit rate requirements will each have to be seriously addressed across all stages of the larger network.Generally a data network that runs on frame relay (FR) technology will not meet the latency criteria that we have specified.There may be some exceptions if the FR virtual circuit involves only two frame switches.The regular Internet is definitely not suitable for corporate VoIP traffic and only a managed net-work can meet the requirements.
A managed IP network is one where the service provider(s) monitor and control the QoS, outside the customers own environment, to mini-mize latency and jitter, while ensuring that sufficient bandwidth is avail-able for busy traffic.In a managed network the VoIP packets pass through the least number of nodes and the QoS is guaranteed, going beyond just best effort engineering.
The characteristics of the access links between the LANs and WAN must also be well defined and managed to fit within the overall criteria budget from source to destination.The topic of virtual private networks (VPNs), which often fulfill this WAN role, is addressed toward the end of Chapter 3, in the Managed Networks section.
Deployment Strategies for IP-Centrex
The arrival of credible IP-Centrex services presents different scenarios to corporate decision-makers, depending on the state of their telecom infra-structures and on the alternative offerings in their marketplace.These situations fall into one or more of three classes:
1. Controlled Migration to IP-Centrex. Many organizations will wish to add IP phones to an existing Centrex configuration (i.e., the hy-brid approach).Two possible reasons for this might be:
◆ To implement a contact center as a managed service;
◆ To extend fully featured, multimedia service to remote or mobile workers.
For these solutions the organization will need to use a well-managed IP network, which may be provided by the ILEC, that currently operates the Centrex service, or may be from another service provider (e.g., an international telco with a wider-ranging network, such as AT&T or BT).
The use of two or more IP-WANs presents a further choice to the customer, as the customer may prefer to have access directly to the competitive long-haul network through a switch at the cus-tomers site, rather than via the IP-Centrex service providers own network access.This alternative is shown in Figure 2.6.
2. Greenfield IP-Centrex Implementation. On the reasonable assump-tion that the average digital PBX needs to be replaced (not just enhanced) after a 10-year life, or that long-term Centrex contracts never last more than a decade, about 10% of all organizations will face a crucial greenfield telephone services decision each year. Some examples of this situation are included in Chapter 5. 3. Move to a Competitive Service Provider. Many organizations obtain
their Internet or VPN connections through an ISP and/or rent Web server hosting facilities from a service provider other than the incumbent telco.In many of these cases this service provider is a good candidate to become the customers communications appli-cation service provider (CASP), in preference to the ILEC.
Video conference terminal Telecommuter Home-based agent Edge switch CLEC-VPN TOIP gateway IP-CTX softswitch ILEC-WAN Customers premises Digital Analog PSTN Trunk gateway IP phones Gateway Gateway Broadband accesses SOHO
When considering whether or not to entrust its in-house voice and video communications to a CASP, an organizations management must evaluate the following topics:
◆ The specific IP-Centrex design being deployed by the CASP; ◆ The match between the feature set available, from the competitive
IP-Centrex service, and that needed by the users;
◆ The local access technology being offered to the WAN and its scal-ability to accommodate growth (or, occasionally, contraction) of bandwidth needs;
◆ The perceived financial stability of the service provider, in a time when the finances of some long-established, large telcos also have been seriously questioned.
