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 maintaining in the signaling path of the entire call for billing and control. Below is a diagram illustrating call flow for an outbound call on our network.
Invite
What starts the call. Call came in from the customer.
100 Trying
VI tells the customer "Ok, we have your request, let see if someone wants the call". VI checks and finds which carrier can route the call. We then send the call to the carrier with an Invite. The carrier sends back a 100 Trying to say they are looking to see if they can route the call.
180 Ringing
The carrier then plays a ring as the destination's phone rings. We pass this packet back to the customer's PBX who plays the ring.
200 OK
Destination answers the call. We send the 200 OK back to the customer, so we can have two-way audio.
ACK
An ACK is passed by VI first, basically tell the carrier "we received your 200 OK". We then pass the 200 OK to the customer whom ACKs our 200 OK acknowledging they received it as well.
RTP Media
Ok, so this isn't a single packet but a ton of packets. 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. Meaning we don't touch the audio.
BYE
In the flow chart the BYE is coming from the termination side, the carrier. We pass the BYE back to the customer, so they can end the call.
200 OK
In this case the second 200 OK is acknowledging the BYE.
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