Panoramisk / The VoIP druid 

Registering a client in the SIP model is not mandatory, a session can be initiated with a peer without requiring a lookup in the SIP directory. But why is this method so important and how does it work? This article explains what is a registration in the SIP world and how it works with Asterisk.
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Automatic network configuration of IP phones is a mandatory feature to any enterprise IP telephony installation. Even if all phones have an embedded web server, allowing its configuration, doing this on dozens or hundreds of phones will be cumbersome and unproductive.

Polycom is part of the mature IP telephony manufacturers and their automatic configuration is really rich and allows configuration setup, firmware upgrade and log collection. This article shows basic samples in order to correctly start in IP phone configuration.
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All SIP phones require that we configure some parameters, at least the SIP server address or name. This server address is used when the phone needs to enter the voice over IP network and registers itself to the registrar¹. The registrar keeps a database of the associated phone with their IP address and the UDP or TCP port they can be joined on, by default the 5060. On top of that mechanism, each call requires some exchanges with a SIP proxy, this one can route the call to its destination or some other proxy server. The proxy location can either be entered in the phone configuration or using a dynamic service that can resolve name versus IP address. Let see how these processes are working and how to implement these.
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1. not the SIP proxy as sometimes stated [↩]