Panoramisk / The VoIP druid 

One of the main issue with SIP is that NAT traversal is a nightmare. When you connect to your corporate network using a VPN, it is not easy to use a sip hard phone. Using a softphone is, in general, the simplest way.

However a small tool really useful can help you: Siproxd
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When we left traditional telephony for a VOIP solution, we have, more than before, the way to connect PBX to all other existing systems. Controlling a phone using his desktop computer is really useful.
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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 [↩]

Since telephony transport is digital, we do transport separately voice and signalling. Each domain should be efficient in different places, performances are different and the way providing service has been adapted on specific architectures and machines. The purpose of this article is to explore how Asterisk fit with regards to these two domains that have been kept for telephony over IP.
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