Ip-To-Ip Routing Application; Stand-Alone Survivability (Sas) Feature - AudioCodes Mediant 2000 User Manual

SIP User's Manual
9.2

IP-to-IP Routing Application

The AudioCodes device supports IP-to-IP VoIP call routing (or SIP trunking). The device
enables Enterprises to seamlessly connect their IP-PBX to a SIP trunk provided by an
Internet Telephony Service Provider (ITSP). The Enterprise can communicate with the
PSTN through the ITSP, which interfaces directly with PSTN. Alternatively, the device can
also provide the interface with the PSTN.
At the same time, the device can also provide an interface with the traditional PSTN
network, enabling PSTN fallback in case of IP network failure. In addition, the device
supports multiple SIP trunks, whereby if a connection to one ITSP fails, the call can
immediately be transferred to another ITSP. By allowing multiple SIP trunks where each
trunk is designated a specific ITSP, the device can route calls to an ITSP based on call
destination (e.g., country code). Therefore, in addition to providing VoIP communication
within an Enterpise's LAN, the device allows the Enterprise to communicate outside of the
corporate LAN, using SIP trunking.
The device interfaces between the Enterprise's IP-PBX and ITSP, allowing SIP trunking
implementation by the Enterprise, for example, in the following scenarios:

VoIP between headquarters and remote offices

VoIP between Enterprise and PSTN through their ITSP
For a detailed explanation on configuring IP-to-IP call routing, refer to the document IP-to-IP
SIP Call Routing Application Note.
9.3

Stand-Alone Survivability (SAS) Feature

The device's Stand-Alone Survivability (SAS) feature ensures telephony communication
continuity (survivability) for enterprises using hosted IP services (such as IP Centrex) or IP-
PBX in cases of failure of these entities. In case of failure of the IP Centrex, IP-PBX servers
(or even WAN connection and access Internet modem), the enterprise typically loses its
internal telephony service at any branch, between its offices, as well as with the external
environment. In addition, typically these failures lead to the inability to make emergency
calls (e.g., 911 in North America). Despite these possible point of failures, the device's SAS
feature ensures that the Enterprise's telephony services (e.g., SIP IP phones or soft
phones) are maintained by routing calls to the PSTN (i.e., providing PSTN fallback).
The device supports up to 250 SAS registered users.
The SAS feature operates in one of two modes:

Normal:
server) to which every VoIP CPE (e.g., IP phones) within the Enterprise's LAN
registers. The SAS agent at the same time sends all these registration requests to the
Proxy server (e.g., IP-Centrex or IP-PBX). This ensures registration redundancy by the
SAS agent for all telephony devices. Therefore, SAS agent functions as a stateful
proxy, passing all SIP requests received from the Enterprise to the Proxy and vice
versa. In parallel, the SAS agent continuously maintains a keep-alive "handshake" with
the Proxy server using SIP OPTIONS or re-INVITE messages.

Emergency: The SAS agent switches to Emergency mode if it detects (from the keep-
alive responses) that the connection with the Proxy is lost. This can occur due to Proxy
server failure or WAN problems. In this mode, when the connection with the Proxy
server is down, the SAS agent controls all internal calls within the Enterprise. In the
case of outgoing calls, the SAS agent forwards them to a local VoIP gateway (this can
be the device itself or a separate analog or digital gateway). For PSTN fallback, the
local VoIP gateway should be equipped with analog (FXO) lines or digital (E1/T1)
Version 5.8
Initially, the device's SAS agent serves as a registrar (and outbound Proxy
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9. IP Telephony Capabilities
October 2009