For conversation purpose, I’m referring to SIP as the use of callcentric.com with a Linksy SPA1001 ATA. That being said, I noticed when calling between ATAs (that are both using callcentric.com) the call quality is superior, so I am venturing that the voice packets are sent directly from one IP address to another, with the least number of hops possible.
I have no way of verifying this with data: if you do, please share. If my hypothesis is wrong, please advise why.
The reason that I am interested is because I am standing up several ATAs overseas. If inter-ATA calling (all using callcentric) is truly Point to Point, then I can set the expectations among users that calls will be of the good quality.
Not sure about callcentric, but callwithus and gigaset.net use only directmedia for free calls between subscribers. This requires forwarding of rtp ports for both parties. Without it you get no audio.
IP to IP is done with Kamalio or OpenSIP( SIP router and Proxy) etc. SIP headers sends data to SIP Router who in turn replies back to to phone with direct ip for UDP and RTP packets to connect directly between phones. Disconnect signiling again goes via a sip notification to the server who disconnects the call. Only sip trafic goes to server which removes a lot of the load.
Asterisk takes in the call process and then send out to device, causing more load because it still processes the call.
So yes VOIP profiders do have a direct connection from one phone to another.
Only SIP signiling is handled by their servers
[quote=“rudio”]So yes VOIP profiders do have a direct connection from one phone to another.
Only SIP signiling is handled by their servers[/quote]
Can have. Most people do not enable directmedia in Asterisk, and I’ve seen reports that some providers will not accept re-invites, to establish direct media.