IP, VoIP & SIP telephony
Telephony is changing and a new generation of Internet-based technologies is delivering lower costs and significant business benefits. Driven by the wide availability of high-speed broadband and the use of Internet protocols in backbone telecoms, many organizations are switching from legacy voice services and analogue/digital handset private branch exchange (PBX) infrastructure.
For many businesses the next step is Internet Protocol (IP)-based telephony and, specifically, use of Voice over IP (VoIP) and the Session Initiation Protocol (SIP).
Benefits of IP telephony
There are a number of benefits to moving to IP-based telephony including:
- Scalability – much easier to add capacity for additional simultaneous calls.
- Flexibility – for multi-site operations, homeworking, easier office moves and phone number transfers.
- Ability to add additional sites and devices and move calls between them quickly and seamlessly to handle overflow, or technical or other failures
- Able to deliver additional functionality such as video conferencing and instant messaging.
- Integration of an organization’s data and phone networks, providing the best use of bandwidth.
- Improved disaster recovery, including highly flexible call forwarding options.
- Easier administration with online management and monitoring tools.
- Future proofing – the telecoms world is moving rapidly to all-IP infrastructure.
How it works
Internet-based telephony is centred around two key technologies: Voice over IP (VoIP) and Session Initiation Protocol (SIP) Trunking.
VoIP is a process of digitising voice calls and then transmitting them across IP-based networks such as LANs and WANs. It is fundamentally different to traditional telephony in that it uses digitized packets of data rather than the old style circuit switching used on the Public Switched Telephone Network (PSTN).
Callers use feature-rich VoIP handsets which contain audio codecs to digitize and compress the voice, and then transmit it as data packets using IP and other protocols to their end destination. At the receiving end, the process is reversed and the data packets are reconstructed back to analogue and delivered to the listener.
There are a variety of handsets available on the market, with an array of features such as speed dialling, built-in directories, caller identification and so on. VoIP phones usually connect to the office LAN using an Ethernet port, but if there is only one cable run per desk they can be paired up with the desktop PC to share the port. An alternative is a handset that works directly through a desktop PC. Many VoIP phones also feature Power over Ethernet (PoE), powering the phone via unused connectors in the Ethernet cabling. This can reduce the need for additional power cabling and sockets and simplifies deployments in certain environments.
The packetization of the voice in VoIP provides the fundamental transportation of a call, but it also needs to be controlled. Session Initiation Protocol (SIP) manages the call, governing its establishment, termination and other elements. It also assists with the process of voice call rendezvous, whereby calling parties can find each other on the Internet using SIP addressing.
SIP Trunks build on this protocol, and VoIP, to deliver virtual telephone lines outside the organization, across the Internet and other, large IP-based infrastructures such as a private MPLS network. SIP Trunking is provided by an Internet Telephony Service Provider (ITSP), who offer a range of call packages and pricing models.
The basic scenario is that a voice call is made from a VoIP phone to an organization’s IP-based PBX using the office LAN and Ethernet infrastructure. If it is an external call, then it is routed out as a VoIP call across the Internet, via the ITSP and SIP trunking, or, where necessary, converted and sent via the PSTN network of a traditional phone company.
Thanks to its inherent flexibility, SIP has wider uses beyond simple voice call management and is also deployed for video calls and ‘presence’ applications (which convey information about a user’s status). Perhaps most importantly, it also provides a firm stepping stone to next-generation Unified Communications (UC) and enterprise application integration.
VoIP and SIP telephony for UK business
activereach’s Internet telephony service features:
- Comprehensive voice call strategy review.
- Advice on VoIP network topologies, traffic management and virtual LAN segregation strategies.
- Design, supply, configuration and support of VoIP phones and handsets.
- Consultancy on SIP Trunking capacity, ROI, configuration and Erlang calculations.
- Full support and advice on transferring from PSTN and PBX/ISDN to IP-based telephony (e.g. number porting).
- Design, supply, configuration and support of IP-based PBX installations .
- Technical advice on IP/PSTN integration, FXS and FXO gateways.
- Fax to email conversion (T38 protocol).
- Comprehensive support for VoIP/IP security and threat protection, such as firewalls, denial of service protection, Intrusion Detection and Prevention systems (IDS/IPS) and Request Flooding.