Configuring Sip Forking Initiated By Sip Proxy; Configuring Call Forking-Based Ip-To-Ip Routing Rules; Call Survivability - AudioCodes Mediant 1000B User Manual

38.4.2 Configuring SIP Forking Initiated by SIP Proxy

The device can handle the receipt of multiple SIP 18x responses as a result of SIP forking
initiated by a proxy server. This occurs when the device forwards an INVITE, received from
a user agent (UA), to a proxy server and the proxy server then forks the INVITE request to
multiple UAs. Several UAs may answer and the device may therefore, receive several
replies (responses) for the single INVITE request. Each response has a different 'tag' value
in the SIP To header.
During call setup, forked SIP responses may result in a single SDP offer with two or more
SDP answers. The device "hides" all the forked responses from the INVITE-initiating UA,
except the first received response ("active" UA) and it forwards only subsequent requests
and responses from this active UA to the INVITE-initiating UA. All requests/responses from
the other UAs are handled by the device; SDP offers from these UAs are answered with an
"inactive" media.
The device supports two forking modes:

Latch On First: The device forwards only the first received 18x response to the
INVITE-initiating UA and disregards subsequently received 18x forking responses
(with or without SDP).

Sequential: The device forwards all 18x responses to the INVITE-initiating UA,
sequentially (one after another). If 18x arrives with an offer only, only the first offer is
forwarded to the INVITE-initiating UA.

To configure the call forking mode:
1. Open the SBC General Settings page (Setup menu > Signaling & Media tab > SBC
folder > SBC General Settings).
2. From the 'Forking Handling Mode' drop-down list (SBCForkingHandlingMode), select
the required mode:
3. Click Apply.
The device also supports media synchronization for call forking. If the active UA is the first
one to send the final response (e.g., 200 OK), the call is established and all other final
responses are acknowledged and a BYE is sent if needed. If another UA sends the first
final response, it is possible that the SDP answer that was forwarded to the INVITE-
initiating UA is irrelevant and thus, media synchronization is needed between the two UAs.
Media synchronization is done by sending a re-INVITE request immediately after the call is
established. The re-INVITE is sent without an SDP offer to the INVITE-initiating UA. This
causes the INVITE-initiating UA to send an offer which the device forwards to the UA that
confirmed the call. Media synchronization is enabled by the EnableSBCMediaSync
parameter.

38.4.3 Configuring Call Forking-based IP-to-IP Routing Rules

You can configure call forking routing rules in the IP-to-IP Routing table. This is done by
configuring multiple routing rules under a forking group. These rules send an incoming IP
call to multiple destinations of any type (e.g., IP Group or IP address). The device forks the
call by sending simultaneous INVITE messages to all the specified destinations. It handles
the multiple SIP dialogs until one of the calls is answered and then terminates the other
SIP dialogs. For more information, see ''Configuring SBC IP-to-IP Routing Rules'' on page
695.
38.5

Call Survivability

This section describes various call survivability features supported by the SBC device.
User's Manual
Mediant 1000B Gateway & E-SBC
758 Document #: LTRT-27055