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Porta
SIP
System Concepts
•
Any of the SDP fields
By default, the following SIP UAs are considered incapable of digest
authentication, so that IP authentication is applied:
• Cisco VoIP gateway (any Cisco gateway running IOS; this does
not apply to Cisco ATA 186/188)
• Nextone SBC
• Sonus switch
• Mera SIP-HIT
• Quintum gateway
• Asterisk gateway
Please ask the PortaOne support team for assistance in adjusting the
information in this table to reflect the desired configuration of your
network.
Understanding SIP Call Routing
When the PortaSIP server has to establish an outgoing call, it must find
out where the call is being sent to. To do this, it will ask billing for a list of
possible routes. In this case the routing configuration is in one central
location, and billing can use information about termination costs to
choose the best route (least-cost routing).
When a call goes through the PortaSIP server, the SIP server may:
• Direct the call to one of the registered SIP clients, if the called
number belongs to the registered agent.
• Optionally, direct the call to the voicemail box (PortaUM
required) if the called number belongs to an account in
PortaBilling, but this account is not currently registered to the SIP
server (is offline).
• Route the call to one of the gateways for termination, according to
the routing rules specified in PortaBilling.
Routing of SIP on-net calls
The SIP server automatically maintains information about all currently
registered SIP user agents, so it is able to determine whether a call should
be sent directly to a SIP user agent.
Routing of off-net calls
You can have different vendors for terminating off-net calls. For example,
you can terminate calls to the US either to AT&T, via a T1 connected to
your gateway in New York, or to a remote gateway from Qwest.
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