Although there are still issues to be resolved to improve voice quality and reliability, voice-over-Internet-protocol offers several advantages over traditional telephone networks, says Johann Madlberger from EMCC Software

Voice-over-Internet-protocol (VoIP) is the transmission of audio data over an IP-based network. This is quite different from traditional telephone calls, where the voice is transmitted over a circuit-switched network.

So what are the advantages of transferring voice over the Internet when there is already a dedicated, reliable network for audio data that has worked perfectly well for more than 100 years?

The most obvious benefit for users is the reduction in costs. Transferring data over an existing Internet connection is usually cheaper than making a phone call, especially for long-distance calls.

Companies and network operators also benefit from reduced costs since they only have to maintain a single (IP-based) network instead of two networks. Additionally, VoIP uses bandwidth very efficiently, further reducing costs for network operators.

Another advantage of VoIP is the simplified integration with other services. The use of one network for all communication services makes it easier to exchange information between these services and integrate them into one application.

For example, a presence service could notify a user that another person is in a meeting. The user could then decide to send an email rather than making a voice call that would not be taken.

Of course, there are also disadvantages when using VoIP.

In contrast to the existing telephone network, the Internet was not designed to transfer real-time data. Therefore it does not provide a guaranteed quality of service.

When making a voice call on the Internet, the audio packets have to compete for bandwidth with other, non- real-time data. In a congested network this can lead to higher latency, more jitter and packet loss, which results in poor voice quality.

Emergency calls can also cause problems for VoIP; in particular, it can be very difficult to geographically locate the caller.

Implementing protocols

There are several technical challenges to overcome when creating a VoIP system. Typically, there are two types of protocols that have to be implemented – a signalling protocol and a media transport protocol. Among other things, the signalling protocol is responsible for registering the client with a server and initiating as well as ending the call. The media transport protocol is used for the transmission of the voice packets.

There are a number of approaches to implementing VoIP. Skype is the best known of the proprietary protocols. Using a standards-based approach, there are two competing choices – H.323 and the Session Initiation Protocol (SIP).

H.323 is a protocol suite defined by the ITU standards body. It is a binary protocol and specifies signalling protocols (H.225 and H.245) as well as media transport protocols to be used by H.323-compliant clients.

Due to its complexity, H.323 is not very widely adopted, and SIP is becoming the signalling protocol of choice for VoIP. The use of SIP within the IP multimedia subsystem has increased its popularity even further.

SIP is a text-based protocol specified by the Internet Engineering Task Force and defined in RFC 3261. It can be used to establish different types of multimedia sessions and is not limited to voice calls. Additionally, SIP provides support for user authentication as well as redirect and registration services.

To send media data over the network, a media protocol has to be implemented. The most widely used protocol is the Real-time Transport Protocol.

To summarise, although some issues need to be solved to provide better voice quality and reliability, it looks as if VoIP is here to stay. In future, it might even make traditional telephone networks redundant.

Johann Madlberger is senior software engineer at EMCC Software

www.emccsoft.com