IP Telephony (IPT) & Voice over IP (VoIP)

Around 1995, a revolutionary advancement in traditional voice networking enabled multimedia traffic, voice/video/data/fax, to be carried over ‘converged’ data networks. The terms Voice Over IP (VoIP) and IP Telephony (IPT) were introduced to describe how circuit switched voice signals were converted into data packets for transport on IP networks.

The two terms VoIP and IPT are generally interchangeable although VoIP is normally used to describe basic voice transmission on IP whereas IPT describes wider PSTN (Public Switched Telephony Network) type services such as voice, fax, DTMF tones etc. on IP.

VoIP Basics
VoIP enables a media conversion device (Media Gateway) to convert voice traffic to / from the traditional TDM (Time Division Multiplex) switched circuit network and the packet switched IP network under control of a call control device (Call Agent). The converged IP network is also capable of carrying voice signalling, management and billing traffic.

Central to the traditional voice switched circuit network is the voice switch which contains in a single unit all the functions necessary for the routeing of voice calls. IPT and VoIP networks distribute media transport, call control and service control functions across separate computing platforms. This separation of functions introduces flexibility in the location, sizing and development of these entities.

Equipment Types
The movement towards voice and data convergence has advanced at a tremendous rate and a number of industry standards and protocols emerged enabling products to be developed for the building of converged networks.

Application Server (AS): provides specialised applications & service logic and interacts with Call Servers and Media Servers using Application Programming Interfaces (APIs).

Call Agent (CA): performs call control function for calls routed via MGs and end clients. It receives and generates signalling messages and translates between ITU E.164 telephone numbers and IP addresses. It can also perform admission control and subscriber screening. Examples of a CA include Gatekeeper (GK), Media Gateway Controller (MGC), SIP Server, Softswitch (SS) or Call Management Server (CMS - used in packet cable networks).

Media Gateway (MG or Media GW): a device under the control of a CA that converts the media (e.g. voice, video, fax, etc) between circuit switched networks and IP networks. It encodes and decodes digitised media signals using a codec (e.g. G.711, G.723.1 G.729, etc.) and performs packetisation and depacketisation to / from IP. The MG also performs echo management, buffers packets using a jitter buffer (in the egress MG only), and can play tones and announcements.

Signalling Gateway (SG): Performs termination and translation of voice signalling channels between circuit switched and IP networks. Provides a bi-directional message switching function between nodes in an SS7 (Signalling System No.7) network and CAs in the IP network. It can be a standalone device or a function integrated within an MG or a Softswitch (in the MG usually PRI and in SS usually SS7).

End User Devices: include IP PBXs (either traditional TDM PBXs with a VoIP gateway function or software based new generation PBX), IP Phones (perform MG function and have an Ethernet connection), PC Clients (such as Microsoft Messenger and Netmeeting).

Other Devices: include Media Servers (MS) such as audio servers or IVRs (Interactive Voice Response), MCU (Multipoint Control Unit) for conferencing, Presence Servers, databases for authentication and directories.

Industry Standards
Around 1996 standards bodies started to ratify early protocol standards that specified how these various entities should communicate and interact with each other for voice and video telephony and conferencing.

H.323 is an ITU-T specification that built on traditional telephony protocols and is the most mature protocol gaining the earliest momentum in the VoIP market. The two main subcomponent protocols are H.225 (call control), & H.245 (bearer control and capabilities exchange). H.225 consists of two main parts Q.931 (basic call control as used in ISDN networks) and RAS (Registration, Admission & Status).

SIP is developed by IETF and is a mechanism to initiate, terminate & modify sessions in an IP network. In June 2002 SIP RFC 2543 was made obsolete when a number of improvements resulted in a group of new SIP RFCs (3261, 3262, 3263, 3264, 3265, & 3266). It has similarities with the Internet reusing HTTP & SMTP, and has a URL addressing scheme. SIP uses a client / server architecture and the protocol is request-response based. It enables personal mobility by tracking down users and delivering calls to an endpoint. It is forecast that SIP shall ultimately replace H.323 as the peer protocol of choice for most IPT & VoIP applications.

MGCP (Media Gateway Control Protocol) & Megacop/H.248 are master/slave protocols used between Call Agent devices and MGs which evolved to satisfy certain deficiencies in the gateway models. MGCP exists as an informational RFC (3435), and Megacop (IETF RFC 3015) & H.248 (ITU Rec.) are the same protocol developed by an IETF and ITU collaboration.

Many other IPT, VoIP & video telephony standards bodies and protocols have been developed to provide specialised communication functions and architectures such as: T.120 protocols for data collaboration; Sigtran IETF Working Group for reliable transport of signalling over IP including Stream Control Transmission Protocol (SCTP) RFC 2960 & for ISDN, SS7, MTP and SCCP over IP; Voice Over Cable CableLabs DOCSIS for underlying hardware and PacketCable for VoIP over cable networks. Other IETF Working Groups include mmusic (SDP, SAP & RTSP), iptel (TRIP), avt (RTP), sipping (SIP-T, NAT & Firewalls), enum (E164 mapping), pint & spirits (PSTN/IP integration), simple (sip for instant messaging & presence).

Implementing VoIP & IPT networks
Carrying real time voice and video on packet switched networks introduces a number of challenges which must be addressed to ensure that the VoIP network elements and the underlying IP infrastructure are engineered appropriately particularly when running multiple voice, data and even video services.

Voice Quality in many cases needs to be equivalent to that achieved on the PSTN. This is a complex area and can be affected by many other factors such as the choice of codec, the transmission delay, packet delay variation (or jitter), packet loss, and echo management.

Quality Of Service (QoS) techniques are used to differentiate, classify, prioritise and police traffic types.

Security measures such as protecting network elements, subscriber data, media traffic and signalling traffic may also be needed.

Regulatory issues can be an issue for many operators who have licenses mandating voice quality requirements such as delay budgets and numbers of Interconnect points to other networks.

Network & Service Migration must be carefully planned and implemented in order to ensure that the end user has no or minimal disruption to service.

VoIP & IPT Future
IPT and VoIP have now established themselves within the voice and data industries. Standards have matured to the point where robust, scalable and reliable products can be readily integrated within existing networks or deployed in greenfield networks.

There is still a significant amount of activity within standards organisations to improve and develop existing protocols particularly in the SIP arena, but this activity is considered a refreshing change to the stagnation that existed with the development of voice services prior to the advent of IPT & VoIP.

Careful design and integration can reduce the risk and delay with deploying IPT and VoIP solutions particularly as every adopter of the technology will have their own requirements with respect to services, legacy infrastructure, operational support arrangements, etc.

Many businesses are increasingly embracing VoIP and IPT as they discover the benefits and opportunities that it can deliver. VoIP and IPT are becoming an important part of many communication strategies and business plans for carriers, enterprises and equipment suppliers as they seek to gain competitive advantage and it is clear is that we have entered a new era in the evolution of voice networking and communications.

Home | Benefits of Convergence | IP Telephony & VoIP | NGN IP Video | Media Streaming | Technology Whitepapers