Technical data

How Calls Are Processed

Dialogic

Diva

SIPcontrol

Software 1.8 Reference Guide page 45

CHAPTER 6

How Calls Are Processed

The Dialogic

Diva

SIPcontrol

Software uses an endpoint-based approach to process calls, which means that

every PSTN interface and every configured SIP peer is considered as a single endpoint. The endpoint saves the

Diva SIPcontrol software settings for the respective PSTN interface or SIP peer. Each call originates at a specific

endpoint (on the SIP side after assigning the SIP call request to one of the configured peers) and needs a route

to find its designated endpoint (the destination). Thus, the most simple configuration needs one PSTN endpoint,

one SIP peer, and one route as shown in red in the graphic below.

This graphic shows that an endpoint is only a virtual object of a real device. The endpoint saves the settings for

the corresponding device. For example, if a call should be routed from SIP device 3 to PSTN device 2 as marked

red in the graphic, then:

• The settings of SIP device 3 need to be configured as SIP peer endpoint in the SIP Peer Configuration,

• the settings PSTN device 2 needs to be configured as PSTN endpoint in the PSTN Interface

Configuration, and

• the condition "called address is 3456" needs to be configured in the Routing Configuration to route the

call to the correct device.

If you have for example a SIP or PSTN device 4 with no endpoints configured in the Diva SIPcontrol software,

then you cannot establish a call, because the Diva SIPcontrol software will not know the settings of the device.

The PSTN endpoint is found via its controller number. On the SIP side, multiple SIP peers may connect via the

same network interface. Therefore, the assignment is more complex:

1. The host/domain name and port number of the received "FROM" header is compared against the SIP peer

settings.

2. If no host matches, the same address is compared against the "Domain" parameters of the SIP peers.

3. If no match is found, the Diva SIPcontrol software looks for a SIP peer with the Default SIP to PSTN Peer

option enabled.

4. If the call cannot be assigned, regardless of whether the call originated in the PSTN or SIP network, the call

is rejected.