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Our call flow is strictly SIP in SIP out. Our carriers interface directly with traditional PSTN and handle converting the calls from SIP to TDM or TDM to SIP. It is VoIP Innovations philosophy that routing the RTP (media) through our switches is unnecessary. It would cause additional network hops, possible points of failure, latency, and jitter. We feel the best voice quality can be achieved by routing the RTP stream to flow directly between our customers' switches and our carriers while remaining in the signaling path of the entire call for billing and control. Below is a diagram illustrating call flow for an inbound call on our network.

 

Explanation:

Invite
Calls come in from a carrier. VI checks and find which customer owns the DID. We then send the call to the your network.

100 Trying
VI tells our carrier "Ok, we have your request, let’s see if someone wants the call". We then pass this to your network.

180 Ringing
The customer then plays a ring as the extension or phone that wants the call and is playing a ring.
VI passes the Ringing packet back to the carrier.

200 OK
Customer answers the call. We send the 200 OK back to the carrier, so we can have two-way audio.

ACK
An ACK is passed by VI first, basically say "we received your 200 OK". We then pass the 200 OK to the carrier whom ACKs our 200 OK acknowledging they received it as well.

RTP Media
This is the audio in data form. If you look at the flow chart it shows a connection between the customer and the carrier. VI isn't in the media stream (we do not proxy the audio).

BYE
In the flow chart the BYE is coming from the originating side, the carrier. We pass the BYE back to the you, so you end the call.
 
200 OK
In this case the second 200 OK is acknowledging the BYE.