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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Posted by: Jean-Samuel Reynaud, on 05/21/2008 | [NAT, proxy, SIP, sip proxy and vpn] | Trackback | Popularity: 22%
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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Posted by: Alexandre Chauvin-Hameau, on 10/19/2007 | [centrex, ENUM, extensions.conf, security, SIP, sip.conf and ToIP] | Trackback | Popularity: 29%
The Canadian manufacturer Aastra will soon officially announce the venue of a new Wi-Fi SIP phone, called 312, even if the net is already buzzing this info. There are already SIP DECT phone at Aastra but since some customers would like to leverage their Wi-Fi infrastructure in order to carry voice, this new product arrive at a good time.
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Posted by: Alexandre Chauvin-Hameau, on 10/17/2007 | [Aastra, IP phone, SIP and wi fi] | Trackback | Popularity: 25%
When working in the IP telephony world it is crucial to know how to use a network analyzer, in order to understand how the traffic is circulating over the network. This article presents some interesting Wireshark features related to voice over IP protocols.
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Posted by: Alexandre Chauvin-Hameau, on 10/12/2007 | [asterisk, canreinvite, codec, IP phone, RTP, SIP and ToIP] | Trackback | Popularity: 100%
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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Posted by: Alexandre Chauvin-Hameau, on 10/10/2007 | [DHCP, DNS, SBC, SIP, sip.conf, ToIP and trunk] | Trackback | Popularity: 47%