
Porta SIP
System Concepts
• After the call is completed, the B2BUA sends accounting
information for the call to the billing.
Terminating SIP calls to a vendor using telephony
PSTN
SIP phone A Phone CGW GW-NY-02
12.34.56.78
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Porta SIP
Porta Billing
• Let’s assume that T1 is connected to Qwest on our gateway GW-
NY-02 in New York, where we are able to terminate calls to the
US. This connection would be described as a PSTN to vendor
connection. The PortaSIP server obtains the address of the GW-
NY-02 gateway in the route information.
• The B2BUA sends an INVITE to the remote gateway (GW-NY-
02).
• GW-NY-02 performs authentication on the incoming call via the
remote IP address. Even if the call was actually originated by A (a
dynamic IP address), but the INVITE request to GW-NY-02
arrived from the PortaSIP server, the PortaSIP’s IP address will be
authenticated. Since PortaSIP is defined as our node,
authentication will be successful.
NOTE: Remote IP authentication on the gateway is not required in this case, but is
highly recommended. Otherwise, someone else might try to send calls directly to the
gateway, bypassing authentication and making such calls for free.
• The call will be routed to the PSTN on the gateway.
• After the call is established, the B2BUA starts the call timer,
disconnecting the call once the maximum call duration is
exceeded.
• After the call is completed, the B2BUA sends accounting
information for the two VoIP call legs to the billing. The gateway
will also send accounting information about the answer/VoIP and
originate/Telephony call legs. The billing engine will combine this
information, since accounting from the SIP server allows us to
identify who made the call, while accounting from the gateway
(c) 2000-2006 PortaOne, Inc. All rights reserved. www.portaone.com
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