The “network-everywhere” methodology has infiltrated our lives and most of us are attempting to negotiate a network connection at some point every day. This is the very nature of Session Initiation Protocol, or SIP, which primarily exists for the purpose of negotiating a voice or video connection between two points.
SIP plays a role in the security industry when it comes to paging systems, intercoms and emergency phones. SIP should not be confused as a standard to ensure the reliability of other security technologies such as voice evacuation and emergency messaging systems. Standards such as UL864, EN 60489A or EN 54-16 apply to these types of systems.
While SIP typically doesn’t fare well when simultaneously addressing multiple zones in evacuation scenarios or otherwise, there are many useful applications in the security space, including connection from phones, desktops and other VoIP devices into paging systems; intercom connectivity to and from help points and emergency distress phones; and simple paging or intercom system protocols without a phone system connection.
SIP is best known in the business world for the simple purpose of signaling phone calls in the IP domain. What is perhaps less often understood is the fact that SIP has little to do with the audio transfer itself. SIP carries out the call setup negotiation between two devices at the initiation phase, signaling call events (transfer requests, for example) during the connection, as well as call termination.
SIP does not play a role in the actual audio/video delivery. In the SIP environment, audio carriage is almost always carried out using RTP (Real-Time Protocol), with parameters and details such as audio formats and transmission methods negotiated between two peers using SIP. The combined power of SIP and RTP can support VoIP as well video connections and audio contribution to monitoring centers, for example. From this perspective, SIP provides a flexible architecture to enable device communication, mainly for point-to-point connections.
Paging systems job one
The majority of SIP-related communications have an IP PBX phone system at the core, such as for paging systems and intercoms. Integrators will require some basic IT and phone system programming knowledge to enable SIP across security applications. This essentially means defining extensions for each connected device through server ports and IP addresses, extension numbers and secrets.
The range of potential connected devices is what makes SIP intriguing in the security space. Linking to paging systems—or alternatively, using an IP phone system as a paging system— simply requires SIP endpoints to translate between IP and analog audio. One example is a Barix Exstreamer device, which feeds its audio output into a traditional paging amplifier that feeds audio to a room, zone or building. Other Exstreamer models will directly feed into speakers with their built-in amplifiers. In both cases, the device supports the necessary audio decoding and represents an IP-addressable, SIP-capable receiving device that the PBX recognizes as a phone extension.
In such scenarios, a security guard in a monitoring center can place a phone (SIP) call direct to the endpoint device and upon acceptance, a relay fires, activating the paging system and, if necessary, overriding general audio streams (such as background music in office environments). Devices with integrated amplifiers override IP-based backup streams and of course do not need the relay to activate.
The advantage for integrators is simplicity, essentially turning an analog paging system or speaker into an IP paging system or IP speaker without additional components.