AudioCodes Mediant 600 - Mapping PSTN Release Cause to SIP Response; IP Destinations Connectivity Feature

SIP User's Manual 282 Document #: LTRT-83310
Mediant 600 & Mediant 1000
Parameter Description
225).
18.4.4 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 a SIP 486 Busy response to IP. If it detects a Reorder tone, it sends a SIP 404
Not Found (no route to destination) to IP. In both cases, the call is released. Note that
if the parameter DisconnectOnBusyTone is set to 0, the FXO device ignores the
detection of Busy/Reorder tones and doesn’t release the call.

For all other FXS/FXO release types (caused when there are no free channels in the
specific Trunk Group), or when an appropriate rule for routing the call to a Trunk
Group doesn’t exist, or if the phone number isn’t found), the device sends a SIP
response (to IP) according to the parameter DefaultReleaseCause. This parameter
defines Q.931 release causes. Its default value is ‘3’, which is mapped to the SIP 404
response. By changing its value to ‘34’, the SIP 503 response is sent. Other causes
can be used as well.
18.4.5 IP Destinations Connectivity Feature
You can configure the device to check the integrity of the connectivity to IP destinations of
Tel-to-IP routing rules in the Outbound IP Routing table. The IP Connectivity feature can be
used for the Alternative Routing feature, whereby the device attempts to re-route calls from
unavailable Tel-to-IP routing destinations to available ones (see 'Alternative Routing Based
on IP Connectivity' on page 283).
The device supports the following methods for checking the connectivity of IP destinations:

Network Connectivity: The device checks the network connectivity of the IP
destination using one of the following methods (depending on the settings of the
AltRoutingTel2IPConnMethod parameter):
• Ping: The device periodically (every seven seconds) pings the IP destination.
• SIP OPTIONS: The device sends "keep-alive" SIP OPTIONS messages to the IP
destination. If the device receives a SIP 200 OK in response, it considers the
destination as available. If the destination does not respond to the OPTIONS
message, then it is considered unavailable. You can configure the time interval
for sending these OPTIONS messages, using the
AltRoutingTel2IPKeepAliveTime parameter.

Quality of Service (QoS): You can enable the device to check the QoS of IP
destinations. The device measures the QoS according to RTCP statistics of previously
established calls with the IP destination. The RTCP includes packet delay (in
milliseconds) and packet loss (in percentage). If these measured statistics exceed a
user-defined threshold, the destination is considered unavailable. Note that if call
statistics is not received within two minutes, the QoS data is reset. These thresholds
are configured using the following parameters:
• IPConnQoSMaxAllowedPL – defines the threshold value for packet loss, after
which the IP destination is considered unavailable
• IPConnQoSMaxAllowedDelay - defines the threshold value for packet delay, after
which the IP destination is considered unavailable