To make VoIP work, it entails hardware and software elements that work together – each with specific functions necessary in the overall scheme of things. The hardware may include the fiber-optic cables and switches that bring broadband connectivity as well as the on-site physical PBX system, local phones and PCs used for configuring the system. The software side may include a hosted PBX system, soft phones, the compression codecs, encryption and packetizing schemes, and the signalling protocols that initiate and end each VoIP call.

Call initiation and teardown

In regular telephone calls, the so-called signalling system 7 (SS7) is the call control protocol that assigns a distinct channel for each PSTN call, providing the information necessary to maintain the call and preventing the line from going dead. SS7 also ensures that call quality is maintained throughout the call.

The high-quality call control in PSTN is one of the earlier barriers that hampered VoIP, as unreliable call mechanisms in the early Internet telephone systems could not offer the same level of quality as regular ‘legacy’ phone systems. Fortunately, new call control methods and advances in dedicated packet-switching enabled VoIP providers to offer services that match if not surpass traditional telephony.

Like in PSTN calls, call control or signalling mechanisms play a central part in VoIP systems, as these manage the overall structure of VoIP calls. The functions of the various signalling methods include determination of the language to be used in all communications for each call, and transmission of the following:

• Message header, phone number or IP of origin, phone number or IP of destination
• Prescribed limit for the number of times the call can be forwarded
• Listing of the accepted commands
• Specialized software for managing the path for the voice portion of the call
• Software for managing the packetization and compression of the media (codecs)

These functions give VoIP the flexibility to offer enhanced call services aside from the standard function of initiating a call and tearing it down once the call is completed. It allows users to connect to multiple endpoints, record the call, route to other endpoints, etc.

Though there are numerous signalling protocols available, most VoIP providers limit the choice to only two options:

Secession Initiation Protocol (SIP)

Currently growing in popularity perhaps due to its similarity to Internet technologies like HTTP, SIP is the de facto standard set by the Internet Engineering Task Force (IETF) for the transmission of media. SIP is structurally faster than other protocols, as it requires only one invite to start a call, and uses a conversational plain text approach. This plain text method allows easy analysis and troubleshooting of VoIP problems.

H.323

This older standard was designed by the International Telecommunications Union (ITU), to support the transmission of both video and voice. The drawbacks of H.323 are its being in code, which makes it difficult for non-programmers to do even simple troubleshooting, and its points have a strict registration process, which means a VoIP phone has to be reprogrammed whenever it is assigned a new number.

Most VoIP hardware in the market, such as phones and gateways, are dedicated to one of the above protocols and hence, cannot be used for both. Even VoIP providers may restrict clients to one of the standards for compatibility concerns. Integration is possible, but calls can only successfully connect after the primary signalling method is agreed upon.

Saiju is an IP communications and hosted business software expert. He has a considerable amount of experience working with hosted VoIP PBX solutions and hosted SIP providers.