Introduction to the semi-digital building Intercom System

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Easywave time: 2014.08.030

Category: Linux applications-semi-digital building Intercom SystemDescription: reprinted. Please keep the link

NOTE: If any error occurs, please correct it. These are my Learning Log articles ......

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I. Introduction to the semi-Digital Intercom System

The so-called half-number refers to the use of TCP/IP protocol for transmission of audio and video, but a digital-to-analog converter must be added between each floor, this digital-to-analog converter is equivalent to an operating system, and also requires an IP address box, which is equivalent to a SIP proxy server, the entire digital transmission part can be transmitted using RTP + sip, this solution solves the problem of audio and video interference caused by transmission distance or other uncontrollable factors during audio and video transmission by the analog intercom system, the transmission part adopts the TCP/IP protocol to solve this problem! The core part of the entire semi-digital building intercom system can be implemented using the Linphone architecture, because the entire Linphone architecture is an open-source network IP videophone system implemented using the RTP + sip architecture. For details, refer to my blog post:

Linux-based open-source VoIP system Linphone [1]

Linux-based open-source VoIP system Linphone [2]

Linux-based open-source VoIP system Linphone [3]

Linux-based open-source VoIP system Linphone [4]

Linux-based open-source VoIP system Linphone [5]

Linux-based open-source VoIP system Linphone [6]

 

An Open-Source SIP server can be used for IP address boxes and can be transplanted to embedded systems. Here I will only talk about a half-digit building intercom architecture, for details about the SIP protocol, refer to the explanation on the network. Here is a brief introduction to the SIP protocol:

• Session Initiation Protocol is a signaling control protocol at the application layer. Creates, modifies, and releases sessions for one or more participants. These sessions can be Internet Multimedia Conferencing [1], IP phone or multimedia distribution. Session participants can communicate with each other through multicast, unicast, or a mixture of the two.
• SIP and Resource Reservation Protocol (RSVP) responsible for voice quality interoperability. It also cooperates with several other protocols, including Lightweight Directory Access Protocol (LDAP) for locating and remote identity authentication for Identity Authentication Dial-In User Service (RADIUS) and RTP and other Protocols responsible for real-time transmission.
• An important feature of SIP is that it does not define the type of session to be established, but only defines how to manage sessions. With this flexibility, it means that sip can be used in a wide range of applications and services, including interactive games, music and video on demand, as well as voice, video, and web conferencing. SIP messages are text-based and therefore easy to read and debug. New Service programming is simpler and more intuitive for designers. The MIME type description is reused as the e-mail client, so session-related applications can be started automatically. SIP reuse several existing mature Internet services and protocols, such as DNS, RTP, And rsvp. There is no need to introduce new services to support the SIP infrastructure, because many of the infrastructure is in place or ready to use.
• The expansion of SIP is easy to define. It can be added by the service provider to a new application without damaging the network. The old SIP-based devices in the network will not impede new services based on the SIP. For example, if the old SIP implementation does not support the method/header used by the new sip application, it will be ignored.
• SIP is independent of the transmission layer. Therefore, the underlying transmission can use an atm ip address. SIP uses User Datagram Protocol (UDP) and Transmission Control Protocol (TCP) to flexibly connect users independent of the underlying infrastructure. SIP supports multi-device function adjustment and negotiation. If the service or session starts the video and voice, you can still transmit the voice to a device that does not support the video, or use other device functions, such as one-way video stream transmission.
  

Ii. Semi-digital building intercom

As shown in, it is a diagram of a semi-Digital Intercom System, which consists of the following parts:

1): intercom host

2): IP address translation box

3): digital-to-analog converter (similar to the switch, which can be implemented using the SIP Protocol)

4): Extension

5): Signal trunk Station

Figure 1: Diagram of the semi-Digital Intercom System

 

 

 

Introduction to the semi-digital building Intercom System