Panoramisk / The VoIP druid 

You are now an IP telephony user and have installed an Asterisk. You are using an IP Centrex link to a SIP or IAX operator. Go to the next level with peer-to-peer telephony with ENUM.
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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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Transporting voice over a packet network is not an easy task and requires controlled quality with regards to bandwidth first but also with delay, jitter and frame loss ratio. We can encounter a very bad voice quality during network congestion, which is not good for a professional network. In order to test our network in advance or during production there are tools and techniques that can be easily used. This article shows how to simply test an IP network for voice transport with off the shelf tools.
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One of the interests of IP telephony is the fact the Internet can become a network to carry telephony for some usages. We cannot consider today this network is offering a high grade quality but its ubiquity is incredible and no telco can nowadays be compared on this topic. It becomes relatively easy to imagine using a foreign ToIP operator in order to access specific prices in this country on one hand and be able to accept calls with a local number on the other hand.
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In order to enable voice call exchange between two PBX the straightforward protocol to use is SIP. It can allows device interconnection, regardless of their respective role, phone, proxy or gateway. This article explains how simply interconnecting two Asterisk PBX and could be used as a template in order to configure a SIP trunk between an Asterisk PBX and any other one available on the market.
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