Determining The Availability Of Destination Ip Addresses; Pstn Fallback As A Special Case Of Alternative Routing; Relevant Parameters; Mapping Pstn Release Cause To Sip Response - AudioCodes Mediant 1000 User Manual

Destination IP address is disallowed if no ping to the destination is available (ping is
continuously initiated every seven seconds), when an inappropriate level of QoS was
detected, or when a DNS host name is not resolved. The QoS level is calculated according
to delay or packet loss of previously ended calls. If no call statistics are received for two
minutes, the QoS information is reset.
7.7.2

Determining the Availability of Destination IP Addresses

To determine the availability of each destination IP address (or host name) in the routing
table, one (or all) of the following (configurable) methods are applied:
Connectivity: The destination IP address is queried periodically (currently only by
ping).
QoS: The QoS of an IP connection is determined according to RTCP statistics of
previous calls. Network delay (in msec) and network packet loss (in percentage) are
separately quantified and compared to a certain (configurable) threshold. If the
calculated amounts (of delay or packet loss) exceed these thresholds, the IP
connection is disallowed.
DNS resolution: When host name is used (instead of IP address) for the destination
route, it is resolved to an IP address by a DNS server. Connectivity and QoS are then
applied to the resolved IP address.
7.7.3

PSTN Fallback as a Special Case of Alternative Routing

The PSTN Fallback feature enables the device to redirect PSTN originated calls back to
the legacy PSTN network if a destination IP route is unsuitable (disallowed) for voice traffic
at a specific time. To enable PSTN fallback, assign the device's IP address as an
alternative route to the desired prefixes. Note that calls (now referred to as IP-to-Tel calls)
can be re-routed to a specific trunk group using the Routing parameters (refer to ''IP to
Trunk Group Routing'' on page 204).
7.7.4

Relevant Parameters

The following parameters (described in ''Routing General Parameters'' on page 198) are
used to configure the Alternative Routing mechanism:
AltRoutingTel2IPEnable
AltRoutingTel2IPMode
IPConnQoSMaxAllowedPL
IPConnQoSMaxAllowedDelay
7.8

Mapping PSTN Release Cause to SIP Response

The device's FXO interface interoperates between the SIP network and the PSTN/PBX.
This interoperability includes the mapping of PSTN/PBX Call Progress Tones to SIP 4xx or
5xx responses for IP-to-Tel calls. The converse is also true: for Tel-to-IP calls, the SIP 4xx
or 5xx responses are mapped to tones played to the PSTN/PBX.
When establishing an IP-to-Tel call, the following rules are applied:
If the remote party (PSTN/PBX) is busy and the FXO device detects a Busy tone, it
sends 486 Busy to IP. If it detects a Reorder tone, it sends 404 Not Found (no route to
destination) to IP. In both cases the call is released. Note that if
SIP User's Manual 416
Mediant 1000 & Mediant 600
Document #: LTRT-83303