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Re: [Sipwitch-devel] Peer-to-peer communication


From: sipwitch
Subject: Re: [Sipwitch-devel] Peer-to-peer communication
Date: Fri, 25 Oct 2013 16:22:52 -0400

> 1) It says SIP witch is peer to peer VoIP server.
> Does this mean that for the initial call establishment
> server would be used and later for VoIP data,
> peer to peer communication is possible between the SIP clients ?

In my opinion, as a new user, SIPwitch manages logging only and when a call is 
initiated by one registered user the Sipwitch instructs VoIP clients like 
CsipSimple or Jitsi or hardware phones to process this call directly. Packets 
will not fly through the Sipwitch itself.
Maybe it is a behavior of a pure Sipwitch. There is a mysterious *-forward 
plug-in but manual is not existent and package is not available for Raspberry 
PI so I can’t tell anything more.


> 2) It says: Calls can be made peer-to-peer behind NAT firewalls,
> and without needing a service provider. What does this mean?

In my opinion it looks like the Sipwitch is able to work at the same subnet of 
the LAN. No metter if behind NAT or not. For example 10.0.0.10 can call 
10.0.0.153 with a netmask 255.255.255.0.
This is a good question what will happen when I use a public IP. Shall it be so 
smart to reinstruct VoIP clients to talk to each other if both are at the same 
LAN 10.0.0.10 - 10.0.0.153 netmask 255.255.255.0 but the server will be for 
example public 200.200.200.200?
Can I call URI address@hidden
What about 2 people at the same private addresses but in different public 
locations?


> 3) Does SIP Witch use hole punching
> mechanism for peer to peer communication?

Good question :-) I will have to try it when I get public and permanent IP 
address from my ISP for some bucks.

I tried to forward port 5060 on the router and CsipSimple registered to my 
Sipwitch behind the NAT without problem. The call itself was impossible. I had 
a ring tone but no voice and no ZRTP yellow bar what is a typical NAT traversal 
problem but the Sipwitch was behind NAT, one CsipSimple was behind the same NAT 
but another CSipSimple was on the public IP through the mobile internet. The 
same mobile internet works with public VoIP servers and ZRTP is set up.

For example ostel.co, ekiga.net and linphone.org servers work with ZRTP in any 
public/non public/NAT/notNAT setup. They always work. For privacy reasons I 
would like to use my own, low power SIP registrar. At least for local calls my 
packets will not need to cross borders and will not be subject to any 
international metadata harvesting.



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