Creating an Open Standards VoIP Community – Session Initiation Protocol (SIP)

Posted by Jeff Csisar on

Session Initiation Protocol, commonly referred to as SIP, is a protocol commonly used in the voice over IP (VoIP) community. The session initiation protocol creates, maintains, and ends data transmission between two or more parties for an established data media stream of voice or video.

What SIP does not do is transmit the media stream itself. SIP works in conjunction with another protocol such as real-time Transport Protocol (RTP) or Secure Real-time Transport Protocol (SRTP) to deliver the actual data.

Why SIP is important is that SIP is the most common open standards protocol for establishing phone calls. Many telecom companies such as Adtran and Aastra use SIP phones and business phone systems to ensure a wide range of interoperability between systems. The famously interoperable Polycom IP SoundPoint phones are a good example of phones that are used by several hosted VoIP companies due to their compatibility with several systems.

Several companies such as Mitel and Cisco use proprietary protocols separate from SIP. While these phones might also support SIP with a firmware flash, they are often limited in features while in SIP mode. Examples of proprietary protocols are Cisco’s Skinny Call Control Protocol (SCCP) and Mitel’s MiNET protocol.

There is no doubt that VoIP for business is the future of the market. While there is nothing wrong with proprietary VoIP protocols themselves, those interested in maintaining a free and open standards internet should consider purchasing an open standards SIP phone and phone system. Open standards make changing SIP trunk providers, Internet Service Providers, hosted PBXs, or even IP phones that much easier.

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