As more and more customers migrate to Voice over Internet Protocol (VoIP) the IT reseller is presented with a new set of challenges to ensure the customer experience meets expectations. VoIP calls are transmitted using packet switching which is fundamentally different to the traditional circuit switching method of transmitting voice data. This introduces a number of opportunities for the reseller seeking to design and implement a network suitable for VoIP:
- Earn additional revenue by offering expert advice on what options are available
- Resell equipment such as phone equipment, network equipment and relevant software
- Configure and install network and VoIP equipment
An understanding of the four key building blocks required to successfully transmit voice data on an internet-protocol network is necessary in order to successfully design and deploy a VoIP solution:
- Codecs
- Protocols
- IP PABXes and IP phones (including softphones)
- VoIP gateways and routers
Codecs A codec is used to encode analogue sound into digital data for transmission and then to decode it again. Each codec has its own characteristics which generally involves a trade-off between quality and bandwidth usage. The most common VoIP codecs are G.711 (used by the traditional phone network) and G.729a which has slightly poorer audio quality. Additionally there are now wideband codecs which allow for a wider dynamic range (think CD versus AM radio) and therefore better audio quality; the most common is G.722. Typical bandwidth requirements for these codecs including typical packet overhead (http://www.bandcalc.com/): G.722 – 95kbps G.711 – 95kbps G.729a – 39kbps Skype and Microsoft use proprietary codecs not listed here. Resellers need to check the capability of hardware phones, softphones, the PABX and the telecommunications provider in order to determine what codecs can be used. It will then be necessary to consider the customer's network connectivity and quality requirements to determine the appropriate codec to use. Protocols There are various signalling protocols which enable VoIP equipment to interoperate and SIP (session initiation protocol) is the most common. A broad range of equipment from many vendors and most of the VoIP-capable telecommunications providers support SIP. Typically SIP-capable equipment from various vendors will operate together successfully although there may be some vendor-specific features that may not work on another vendor's equipment. In addition to the signalling, the actual voice data is typically carried using RTP (real-time protocol) which has an associated protocol RTCP (real-time control protocol). The signalling data may be carried as cleartext or may be encrypted using TLS. Similarly the audio data may be encrypted using SRTP although this is less common currently due to complexity with encryption keys. Encryption of either the signalling or audio data requires support by the equipment and the telecommunications provider. Note that once the call passes to the normal telephone network (PSTN) it will probably no longer be encrypted. IP PABXes and IP phones (including softphones) There are a range of vendors offering VoIP telecommunications equipment (including on-premise and hosted options) and software. The IP PABX choice will typically be a question of price versus performance although integration with existing business systems is becoming increasingly important. End-user equipment is very much a personal choice with some customers requiring dedicated hardware VoIP phones while others prefer to use softphones or to install software on their smartphones and use the Wi-Fi network. In the case of wireless connectivity, resellers need to carefully design the wireless network to deliver the connectivity required for successful VoIP calls. If portability is important, the reseller should consider the use of DECT technology which is an excellent choice for cost-effectively providing good quality calls over a wide area. VoIP gateways and routers A well configured network is vital in order to provide the customer with reliable, good quality VoIP calls; the router is the key component in that network. Older routers and consumer-grade or low-end business grade routers are frequently inadequate to deal with VoIP data and an upgrade to a business-grade router is highly recommended. SIP presents a number of NAT-traversal challenges which will require some network expertise to avoid one-way or no-way audio. Any firewall devices will need to be configured to avoid blocking VoIP traffic (which will typically be carried on a range of ports). It may be appropriate to segment a network into separate VLANs for voice and data to avoid large amounts of non-voice data traffic adversely impacting voice quality. It may also be appropriate to enable traffic prioritisation which gives priority to time-sensitive network traffic (such as voice data) by configuring quality of service (QoS) settings on network and telephony equipment. Note that most Internet connections will not offer any QoS so be careful not to 'oversell' the benefits. The three common VoIP protocols all need to work together in order for a successful VoIP call to occur. The packets may be carried as UDP (more commonly) or TCP packets over an IP network. In the case of UDP, the reseller will need to take care to configure network devices appropriately because networks are typically well-tuned to handle web-traffic, email etc which is carried over TCP. For example the UDP session timer may need to be adjusted to avoid the NAT closing and preventing inbound calls routing correctly. Where on-premise voice equipment needs to connect to a telecommunications provider or a hosted PABX, a WAN connection will be required. Typically this will be an Internet connection so the reseller must match the customer requirements with the budget whilst carefully setting expectations. Customers typically view download speed and data allowance as the key criteria in selecting an iwnternet connection. However in the case of VoIP, upload speed, latency and packet loss are more important. Many businesses successfully use an ADSL connection for VoIP although the 'best efforts' nature of the connection will result in call quality issues on occasions. The typical upgrade path would be to an SHDSL connection and then to fibre-optic. It is quite common to put standard Internet traffic on an ADSL (or VDSL) connection with time-sensitive traffic on SHDSL. This provides a cost effective way to satisfy the requirements for web traffic (high download speeds and large data caps) and VoIP (high reliability with synchronous connectivity). Conclusion VoIP call quality can and should be at least as good as that of the traditional phone network. Resellers should take the time to explain the differences between VoIP traffic and other non real-time data so that customers are prepared to make the necessary investment that will deliver the quality and reliability they expect.
