Experimenting with New Methods in Voice over IP

Being able to integrate voice and data is a strong desire of business as well as of residential users. This type of integration is inefficient, however, on the telephone legacy of channelled voice slots having time division multiplexing (TDM) with expensive associated equipment. A data network like the Internet, with its statistical TDM, uses precious bandwidth much more efficiently. Moreover, the universal presence of IP both in wide area and local area networks makes it a convenient platform to launch VoIP traffic. In this article we share our experience of testing VoIP in our Linux lab. We discuss the entire VoIP environment, including the soft switch and the telephony hardware needed to improve voice quality.

There is much more to IP telephony than simply being able to talk over Internet. To be successful, an IP telephony system must provide all the facilities that are provided by modern public switched telephone networks (PSTN). The quality of service (QoS) for an IP telephony system also should be comparable to that of PSTN. For the deployment of IP telephony, the infrastructure--that is, an IP network--is already available in many cases. It is required to implement protocols necessary for signaling, media transfer, QoS, as well as various features, such as voice mail, billing and son. Unlike PSTN, IP telephony has computers at their endpoints, which are extremely powerful. This allows much of the power and intelligence of the system to be concentrated at endpoints.

We have performed some fundamental experiments on VoIP in the Computer Communications Lab at the University of Engineering & Technology (UET) in Lahore, Pakistan. Our emphasis is on the applications and performance considerations for IP telephony. We are using Quicknet's voice telephony cards for our telephony clients. We have based our VoIP network on VOCAL, which is a soft switch from Vovida Networks. VOCAL enables advanced telephony features, applications and services to be effectively deployed on converged datacom-telecom networks. In addition, we are using ohphone, which is an H.323 client, and a PSTN gateway, both of which are from the OpenH323 project. All these applications run on Linux. Below we discuss the benefits that Linux provides when is used as a VoIP platform.

VoIP protocols are mainly divided into signaling and media protocols. Signaling means call setup, teardown and control. As the number of features and services grow, signaling becomes equally challenging. Unfortunately, a single signaling technology has not yet been defined. But Session Initiation Protocol (SIP), H.323 and Media Gateway Control Protocol (MGCP) are the front-runners. All these protocols are undergoing rapid developments in their capabilities and features. Currently, users of different signaling techniques cannot generally make calls to each other. However, a protocol translation between different signaling protocols can be used in this situation. We discuss a SIP-H323 translator below.

Regardless of the signaling protocol used to setup calls, there appears to be no controversy over the use of the Real Time Protocol (RTP) for media transport. The Real Time Control Protocol (RTCP) is an associated protocol, which provides end-to-end monitoring of data delivery and quality of service. They are independent of underlying transport and network layers. Most commonly RTP and RTCP are used on top of UDP.

Figure 1: SIP Distributed Architecture

SIP is an IETF standard and is described in RFC-2543. Although H.323 is in widespread use in the VoIP market, it is not necessarily the best available protocol. Its competitor, SIP, has a number of advantages that makes it a superior choice for VoIP applications. The first is advantage is SIP focuses exclusively on telephony issues, originally designed to deal with telephony as opposed to other protocols. H.323 and MGCP both support telephony functions in a manner similar to that of phone companies on their circuit switched networks, which is not the most efficient method on a packet-switched network. SIP's distributed architecture handles load surges and service interruptions efficiently in the distributed model of IP communications.