SIP VoIP

The SIP VoIP protocol is one of the most common protocols used in VoIP equipment. SIP-based VoIP has been described as "plain vanilla VoIP."

One common misconception among those new to SIP VoIP is that SIP is used to transmit voice. In fact, SIP is only used to initiate, modify, and terminate connections. Once a connection is established, SIP is no longer used. Instead, a codec (including the common G.711 codec) is used to compress voice entering the system, transmit it as data packets, and decompress that voice for playback on the other end of the call.

One key advantage of the SIP protocol for VoIP, which has likely contributed significantly to its widespread popularity, is its flexibility. SIP can be enhanced with new versions, while still maintaining backward compatibility with older versions. SIP 2.0 is a very common SIP version, as it improved upon SIP 1.0 while maintaining compability.

When selecting VoIP gear, it's important to look like a common protocol like SIP. This offers excellent flexibility later, so you're not trapped with the same manufacturer in the future.

Equally important is selecting a non-proprietary, industry-standard codec, for the same reason as protocol. If you choose one manufacturer's proprietary protocol, you'll be hard-pressed to find compatible equipment from any other vendor.

For an example of a VoIP solution based on industry-standard protocols and codecs, look at the VoIP order wire system from DPS Telecom. This communications system, which can optionally be integrated with the company's RTU line, uses SIP 2.0 and G.711 for voice communication. Plus, integration with RTU's helps you roll two purchases into one. This also saves rack space, because you only need 1 RU for both remote monitoring and VoIP communications for your technicians.