Voip Via proxy

[Holt]

At home can call from internet but at hostel no call. Is it possible that I am using a PC that I use as a proxy to call from my pc? Some students from my room set the router in the settings obtained from a home PC as a voip server running? Would this work?
thank you .

[timon]

you can also imagine a skype phone that you connect to your PC and you can even in my free for a cheap rate local call. Skype 4.0.0.244

[Steve123]

You must use an Internet line and do not have a phone line, otherwise you would also not a normal phone directly to your modem to connect. But if there is a major user within your home, you also might consider a "home" plant. On the market are those things that a small squad, and you do as your line and then turn all phones on which you want without you their line.

[Holt]

I want Parental House Voip server or something set up, and then the modem on my student set voip via the parental home

[Zachary]

Here are the list of Free VoIP Proxy that you can use with your VoIP System:

Partysip
Partysip is an implementation of a SIP proxy server. SIP stands for the Session Initiation Protocol and is described by the rfc2543 (soon to be deprecated by latest revisions). SIP is a open standard replacement from IETF for H323.

Partysip is a modular application where some capabilities are added and removed through plugins. The program comes with several GPL plugins. At this step, partysip and its plugins could be used as a 'SIP registrar', a 'SIP redirect server' and a 'SIP statefull proxy server'. (stateless capabilities have been removed)

siproxd - SIP proxy/masquerading daemon
Siproxd is a proxy/masquerading daemon for the SIP protocol. It allows SIP clients (like kphone, linphone) to work behind an IP masquerading firewall or router
.

Load Balancer Proxy
The Load Balancer is a very simple proxy that is useful in SIP-based VoIP installations where there are multiple ingress proxy servers. The Load Balancer permits pooling these servers, thereby eliminating the need to balance user demands for connectivity through a complicated provisioning algorithm.

All users can send their INVITEs and REGISTERs to the same SIP URI and the Load Balancer will assign ingress proxy servers dynamically to each transaction. In this way, the traffic load is balanced over a pool of proxy servers based on the real-time demand for services.

STUN Server
The STUN (Simple Traversal of UDP through NATs (Network Address Translation)) server is an implementation of the STUN protocol that enables STUN functionality in SIP-based systems. The STUN server tar ball also include a client API to enable STUN functionality in SIP endpoints. In addition there is a command line UNIX client and a graphical windows client that check what type of NAT the user is using.
STUN is an application-layer protocol that can determine the public IP and nature of a NAT device that sits between the STUN client and STUN server. The current version of the code supports most of RFC 3489 except the ability to get OTPs from the server.