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1 Voice over IP (VoIP) for Telephony Advantages of VoIP Migration for SMBs BLACK BOX Hybrid PBX VoIP Gateways SIP Phones Headsets blackbox.com

2 Table of Contents Introduction...3 About Voice over IP... 3 VoIP Considerations...3 Delay or Latency...3 Jitter...3 Pulse Code Modulation and Voice Compression...4 Echo Cancellation...4 Packet Loss...4 Transport Protocols...4 Dial Plan Design...4 Security...4 VoIP Pros and Cons...5 Protocols of VoIP...5 How VoIP Works...5 Phones...6 Today s VoIP Network...7 How to Calculate Bandwidth...7 How to Select a VoIP Solution...8 Cost Considerations...8 Serviceability...8 Infrastructure...8 How to Implement VoIP...9 Our Perspective...9 About Box...9 We re here to help! If you have any questions about your application, our products, or this white paper, contact Box Tech Support at or go to blackbox.com and click on Talk to Box. You ll be live with one of our technical experts in less than 30 seconds blackbox.com Page 2

3 Introduction The existence of broadband Internet has enabled the introduction of Voice over IP (VoIP) services, as an alternative to expensive call handling over the Public Switched Telephone Network (PSTN). VoIP enables subscribers to initiate and receive telephone calls in the same manner as PSTN. However, VoIP doesn t stop at that; it has evolved into a complete set of telephony capabilities that could surpass today s traditional telephony. Although VoIP was planned to be the solution to eliminating long-distance call charges, it was subject to qualitative hurdles before becoming reality. System costs and Quality of Service (QoS) have been the challenges for VoIP. This document will describe VoIP, its implementation and the challenges. Box s IP PBX phone systems offer interoperability gateways between PSTN and VoIP networks About Voice over IP Private Branch Exchange VoIP is a suite of communication protocols whose main purpose is the delivery of voice communications and multimedia sessions over the Internet by using the Internet Protocol (IP) suite. Voice is digitized using codecs and is transmitted inside IP packets just like any other data. The Internet was originally built for data transfer relying on best-effort delivery of data packets as fast as possible. This introduced real-time constraints to support PSTN-quality telephone calls. Voice packets however, can be identified and recognized as real-time packets using QoS mechanisms. This creates a host of considerations before implementing a VoIP solution. VoIP Considerations Delay or Latency VoIP delay or latency is measured as the amount of time it takes for speech to exit the speaker s mouth and reach the listener s ear. The two main delays in telephony networks today are propagation delay and handling delay. Propagation delay is caused by the duration that a signal (light or electric pulse) requires to traverse a medium such as a fiber or copper-based network. Handling delay is the processing delay, and it encompasses many different causes of delay such as packetization, compression, switching, etc. Handling delay is caused by devices that intercept and forward the frame throughout the network. Box s IP PBX phone systems offer G.729a, G.711 u-law/a-law, and GSM compression algorithms. Queuing delay is a result of packets being held in a queue because of congestion on an outbound interface when more packets are sent out than the interface can handle at a given time interval. For optimal performance, this factor should be less than 10 msec. The G.114 recommendation specifies that for good voice quality, no more than 150 msec of one-way, end-to-end delay should occur. Jitter Jitter is the variation of packet interarrival time. As an example, the sender is expected to transmit voice packets at a regular interval of 20 msec. These packets can be delayed throughout the packet network and might not arrive at the same regular interval 20 msec at the receiving end. Jitter is the difference between when the packet is expected and when it is actually received. Box s IP PBX phone systems implement a jitter buffer to conceal interarrival packet delay variation. This jitter buffer is adaptive in that it adjusts its size to target a predetermined, allowable late-packet ratio. To compensate for the unpredictable nature of the packet network, the more jitter there is in a network, the larger the jitter buffer needs to be. A jitter buffer is advantageous in a not-so-optimal local area network blackbox.com Page 3

4 Pulse Code Modulation and Voice Compression Pulse Code Modulation (PCM) is the formal name for G.711 ITU-T standard for audio compression or decompression. It uses a sampling rate of 8000 samples per second. Each voice packet is quantized, with 8 bits, resulting in a 64-kbps rate. There are two variations of G.711: u-law and a-law. Historically, North America uses u-law, and Europe uses a-law. It is important to note that when making a long-distance call, any required u-law-to-a-law conversion is the responsibility of the u-law side. Box s IP PBX phone system supports both these variations. PCM is the leading modulation in most of today s phones. Echo Cancellation Echo is the phenomenon of hearing one s own voice in the receiver after a delay of more than 25 msec. It causes interruptions and breaks the cadence of a conversation. In packet-based networks, echo cancellation can be achieved, by integrating echo cancellers into low-bit-rate codecs and operate them on each Digital Signal Processor (DSP). Some vendors implement echo cancellation through software. Box has implemented echo cancellation by introducing a dedicated echo cancellation integrated circuit, hence allowing for a Telco-grade voice quality. Such echo cancellation results in a superior voice quality. Packet Loss Packet loss is a reality that is both common and expected in data networks. When it comes to voice transport, it is important to build a network that can successfully transport voice in a reliable and timely fashion. Also, it is recommended to use a mechanism to make the voice resistant to periodic packet loss. For that purpose, QoS tools that enable administrators to classify and manage traffic through a data network are needed. Transport Protocols The two main types of transport protocols that ride upon IP are the User Datagram Protocol (UDP) and Transmission Control Protocol (TCP). UDP is used when simplicity and reliability are the main concern. Real-time Transport Protocol (RTP) was specifically tailored for delay-sensitive traffic and rides on top of UDP. There, VoIP is carried with an RTP/UDP/IP packet header. Box s IP PBX phone systems rely on RTP for carrying voice traffic. Dial Plan Design Designing a dial plan is one of the biggest challenges in designing an enterprise telephony network. Most companies must decide on their dial plan based on several factors: plans for growth, cost of leased circuits, cost of virtual private networks (VPN), cost of additional equipment for packet voice, number overlap, call flows, etc. Box s phone systems simplify creation of dial plans by implementing a pattern generator, which helps a user build the calling patterns that are used to govern how the system controls outbound calling. The phone system can add or delete digits according to an assigned rule before sending the digits to the carrier. Outgoing call rules could contain one or more dialing patterns. If the number dialed matches any of the dialing patterns, the call will be routed accordingly. As a user example, an organization may choose to route local calls through the analog trunk and long-distance calls through the VoIP trunk. Any dial attempt that does not match a dialing pattern within the given rule will be checked against extensions or features; otherwise the call fails. This applies to every call made from a phone connected to a Box phone system. Box s pattern generator makes dial plans easier than they once were. Security Security of VoIP has become a concern for new adopters over the recent years. Most firewalls do not support VoIP except through VPN connections. Data firewalls, by default, will probably prevent the operation of VoIP by users on the untrusted network when they call devices on the trusted network. Encryption has been the trend for VoIP security. Once encryption choice has been implemented, encryption functions must be implemented in the IP phones, gateways, and call servers blackbox.com Page 4

5 VoIP Pros and Cons Pros Makes a call virtually anywhere that Internet access is provided PBX cost savings Call-time cost savings (reduced long-distance charges) Gain IP-adopted features that are not available from PSTN Increases profitability, lowers the cost of doing business Increases your competitive advantage Cons Security issues inherited from IP IP phone costs are high while each extension requires a new phone High IP bandwidth requirement Delay and echo considerations Traffic QoS requirements Protocols of VoIP There are several protocols that play a role in voice delivery over IP. This document will describe two main protocols. The first is session initiation protocol (SIP). SIP is used for call control such as creating, modifying, or terminating a call, whether this call is a unicast or multiparty. It is a signaling protocol that is equivalent to SS7 or higher layer signaling. SIP allows for voice call sessions, videoconferencing, instant messaging, presence information, file transfer, streaming of multimedia, etc. In the simplest configuration it is possible to use just two SIP User Agents (UA) that send SIP messages directly to each other. However, a typical SIP network can contain a multitude of basic elements: user agents, proxies, registrars, and redirect servers. Box s phone system plays many of these roles in a SIP network environment. Another protocol that plays a vital role in voice delivery is RTP. RTP is the technical foundation of VoIP as it defines a standard for delivering audio and video packets. SIP and RTP are among a host of protocols that Box incorporates to provide excellent delivery of voice and video. How VoIP Works An IP phone boots up just like any other peripheral on a Local Area Network (LAN). Once the bootup process is complete, the IP phones and/or gateways register with a SIP registrar, also known as a call server. The IP phone and/or gateway must first access a Dynamic Host Configuration Protocol (DHCP) server to obtain an IP address. The DHCP server may be part of the data network, it may be a separate server, or it can be integrated within the call server. Once an address has been assigned, the IP device contacts the SIP registrar to register. A Box phone system acts as a SIP registrar and DHCP server in this capacity blackbox.com Page 5

6 The call server may have a common set of privileges and restrictions for IP devices, or an administrator can make the feature assignments. The call server or another assigned server also adds this device and its phone number(s) to the Domain Name Server (DNS) to support directory services. When a user picks up the phone, the dial tone can be generated locally by the phone or by the call server ( Box phone system). The IP device then sends one or more packets requesting a connection and the features to be implemented during the connection, such as a conference call. The call server then determines whether the other device is available or busy. If available, the call server contacts the receiving device and instructs both the caller and called devices to establish a peer-to-peer UDP path to carry the RTP speech. The call server becomes dormant during the call, until one of the devices terminates the call. The call server then breaks the peer-to-peer connection and records the call event as part of the Call Detail Record (CDR). It is important to note here that a call server is not the telephone switch. It controls all the services offered, provides control over the call, supports the telephone features, authenticates and authorizes the caller, and implements security. The speech packets are passed directly from phone to phone while the call server is there to process the signaling and not to switch the speech. It is worth noting here that the Box IP phone system (IP PBX) can operate as a call server and switch at a single point of time. An important element of today s VoIP implementation is the gateway to the Wide Area Network (WAN). When implementing VoIP, you can keep your current access method to the PSTN or your carrier. For that purpose, a trunk gateway is required. A trunk gateway could have an analog (FXO) or TDM (T1/E1) connection to the carrier. Box s IP phone systems support trunk gateway functionality. Phones In today s VoIP networks, there are two main options for phone selection. The first option is using the existing traditional PSTN phone. This option is an attractive choice for first-time VoIP implementers because there is no cost incurred from the purchase of costly IP-based handsets. For that purpose, an analog telephone adapter (ATA) is used to interface with the VoIP network. The user can experience limited capabilities of the attractive set of telephony features available over IP. IP phones could include a host of capabilities leading to high costs. As a matter of fact, VoIP phones represent the largest investment cost when it comes to deploying a VoIP network. Such phones usually contain a DNS client, Session Traversal Utility for Network Address Translation (NAT), DHCP client, signaling stack such as SIP, RTP stack, various codecs for both audio and video, and a user interface to manage all the mentioned items. An IP phone can connect directly to an IP network using wired technology such as Ethernet or wireless such as Wi-Fi. Note that such phones require electric power, whether Power over Ethernet (PoE) or direct power source, unlike traditional PSTN phones. Box's IP phone systems communicate successfully with a host of phones from various vendors blackbox.com Page 6

7 Today s VoIP Network In today s networks, LANs and WANs can support VoIP operations. However, there is a significant difference between LAN and WAN performance that affects signaling speed and consequently voice quality. LANs use Ethernet as a transport medium and operate at 10, 100, or 1000 Mbps on networks with very short delay, no jitter, few errors, and virtually no packet loss. VoIP traffic could share the LAN with data users. It is recommended, nevertheless, that the voice and data devices run on separate VLANs on the LAN switches for both performance and security reasons. Box s voice quality and signaling execution speeds are as good as a traditional TDM PBX. WANs present a number of performance obstacles: limited bandwidth, end-to-end delays, jitter, and packet loss. Extra bandwidth and QoS techniques can solve these problems. Voice calls consume bandwidth about 80 Kbps when no voice compression is used (G.711) and about 25 Kbps when voice is compressed (G.729). If bandwidth is not increased on the WAN, then voice and data users will suffer call degradation. Today s VoIP network is largely enabled by SIP. A typical small-to-medium business network still connects to PSTN using various methods such as T1/E1, PRI, FXO, SS7, DSL, or cable. Although VoIP has picked up momentum in recent years, traditional phones are still widely used. An average office telephony network contains a PBX, a LAN switch, and a WAN router gateway. How to Calculate Bandwidth Several factors contribute to bandwidth calculation: sampling, quantization, encoding, and compression (optional). There is a trade-off between the most prominent codecs in the market today: G.711 PCM and G.729a. G.711 PCM requires a bandwidth of 64 kbps. It has low complexity and consequently, minimal processing power. G.729a, which replaced the original G.729, requires a bandwidth of 8 kbps. It s characterized with medium complexity and typically increases the cost of the system of choice. As an example, let s take G.711 PCM as a compression method. Assuming a sampling size of 20 msec and an Ethernet medium, you can calculate the required bandwidth. Codec payload (bit rate x sample size) 64 kbps x 20 msec 160 bytes Layer 2 overhead (Ethernet header plus voice VLAN tag header) 18 bytes + 4 bytes 22 bytes Layer 3 overhead (IP header) 20 bytes 20 bytes Layer 4 overhead (UDP and RTP) 8 bytes + 12 bytes Total Packet Size Packets per second (pps) 1000 msec / 20 msec 20 bytes 222 bytes 50 pps Total Bandwidth (50 pps x 222 bytes) x 8 bits 88.8 kbps blackbox.com Page 7

8 How to Select a VoIP Solution Selecting a VoIP solution for an organization encompasses an array of decision making regarding cost and serviceability. Cost Considerations Small businesses often don t have the same options in adopting new technologies as large businesses do. Funds and resources restrict decision making for small businesses. VoIP can offer significant cost reduction and a number of other features that are otherwise unaffordable. Traditional phone services cost a small business around three times more per employee. VoIP service providers recognize this emerging market in SMB and now offer packages tailored to SMBs. The initial investment in VoIP remains the most important challenge. Box's VoIP PBX offers a wide variety of features with no additional costs or license fees. There are four cost sources: IP phone terminals Core infrastructure for call handling Service License fees (ongoing) IP phones today are the most expensive element of the overall solution. However, many vendors are emerging with cheaper alternatives that fit the bill for small business. Box s phone systems are compatible with a wide array of phone options. The core infrastructure for today s VoIP is heading towards a bundle offering, where one box contains multiple call-handling elements plus inherited features from a traditional PBX. An IP PBX, such as Box s, could provide an optimal solution that is cost effective and efficient. Service providers might offer an overall bundle that includes the core infrastructure. There are also service integrators who, along with service providers, could present cost-effective solutions that include the terminals. Serviceability It is recommended that an organization studies the services that can be made available through a VoIP offering. An organization can have the flexibility of hiring people to work from home because of the ability to handle calls over a VPN tunnel. In addition, an organization can have more employees on the road at a low cost with the availability of softphones. The one major benefit of VoIP is the ability to move around employees without the extra effort of relocating phone extensions. Those examples highlight the flexibility and mobility that VoIP can bring to an organization. Nevertheless, VoIP brings a new dimension to network management that didn t necessarily exist before. Infrastructure Depending on the state of the data network and telephony structure in a company, a business infrastructure can be dramatically changed from the introduction of VoIP. VPN concentrators might be required to handle remote and mobile soft clients. New firewalls that handle SIP traversal over NAT might be a necessity. The IP network planning could require an overhaul because of new IP assignments blackbox.com Page 8

9 How to Implement VoIP Upon selection of a VoIP solution, the implementation task is crucial for success as it might affect scalability and productivity of the solution. Data network disruption should be reduced to minimum and network redundancy should be the main concern. One approach would be the reorganization of the data network as a first step before introducing voice services. Another approach could be the introduction of VoIP internally in an organization before moving to full-scale VoIP service using a VoIP provider. Size and network structure affect the decision making. Our Perspective VoIP creates an expanded role for the network manager. Not only will LAN deployment and troubleshooting continue to exist, but there will be new responsibilities such as VoIP protocol operation, call server interaction, network performance optimization, and IP phone configuration. Box offers a complete PBX system that is easy to use and manage. This highly scalable IP PBX phone system makes communications easier and more cost effective for small businesses. IP PBX systems are still relatively out of reach for many small businesses because of high cost. Box makes this technology affordable for the small-to-medium size business, by building on the flexibility, performance, and cost savings made available by VoIP. From details on our Hybrid PBX VoIP Gateways, go to blackbox.com/go/voipbx. About Box Box Network Services is a leading voice communications solutions provider, serving 175,000 clients in 150 countries with 200 offices throughout the world. The Box catalog and Web site offer an extensive range of products including headsets, SIP phones, and united communications and conferencing options for small and medium businesses. More information is available at blackbox.com/go/voipbx. Box is also known as the world s largest technical services company dedicated to designing, building, and maintaining today s complicated data and voice infrastructure systems. Copyright All rights reserved. Box and the Double Diamond logo are registered trademarks of BB Technologies, Inc. Any third-party trademarks appearing in this white paper are acknowledged to be the property of their respective owners. WP00056-VoIP-Telephony_v blackbox.com Page 9

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