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1. Introduction
These Application Notes describe the procedure for configuring the AudioCodes MP-118 Analog
VoIP Gateway with Avaya SIP Enablement Services (SES) and Avaya Communication Manager.
The AudioCodes MP-118 Analog VoIP Gateway serves as a gateway between legacy analog
endpoints/trunks at a branch location and a VoIP infrastructure at a main location using the Session
Initiation Protocol (SIP). The MP-118 has 4 FXS (analog endpoint) ports and 4 FXO (POTS trunk)
ports. The compliance test focused on the interoperability of the FXS ports. The FXO ports were
configured only as a failover path to the PSTN if the data WAN is unavailable and SIP calls can not
be made. The ability of these FXO ports to provide local PSTN access for the branch as part of
normal operation is not covered by these Application Notes. The data WAN being unavailable and
causing a failover can be due to three types of failures at the branch: a failure of the WAN link, a
failure of the LAN providing access to the WAN or power loss of the MP-118. In the case of power
loss, the FXS ports are connected directly to the corresponding FXO ports (i.e., FXS port 1 is
connected to FXO port 1, FXS port 2 is connected to FXO port 2, etc.).
The MP-118 registers with the Avaya SES as a SIP endpoint for each analog endpoint connected to
it. When a call is placed from an analog telephone, the MP-118 will send SIP signaling messages to
the Avaya SES to setup the call. Once the call has been setup, the MP-118 converts the analog
signal from the analog telephone to a series of voice samples sent in data packets over the data
network using the Real Time Protocol (RTP).
1.1. Configuration
Figure 1 illustrates the configuration used in these Application Notes. In the sample configuration,
two sites are connected via an IP network. The main site has an Avaya SES and an Avaya S8300
Media Server running Avaya Communication Manager in an Avaya G350 Media Gateway.
Endpoints include an Avaya 4600 Series IP Telephone (with SIP firmware), an Avaya 4600 Series
IP Telephone (with H.323 firmware), an Avaya 6408D Digital Telephone, and a fax machine. An
ISDN-PRI trunk connects the media gateway to the PSTN.
The branch site has an AudioCodes MP-118 Analog VoIP Gateway with two analog telephones, and
a fax machine. The branch site also has two Avaya 4600 Series IP Telephones (with SIP firmware).
The MP-118 connects the branch site to the PSTN via an FXO (POTS) trunk which is only used if
the data WAN is unavailable. The other three FXO ports were not connected. Normally, all
outbound calls from the branch location pass across the data WAN, through the Main site and reach
the PSTN via an ISDN-PRI trunk at the Main site. Conversely, all inbound calls for the branch site
come from the PSTN across a trunk connected to the Main site then passed over the data WAN to
the branch.
All SIP telephones and analog telephones at both sites are registered directly to Avaya SES and are
administered as Outboard Proxy SIP (OPS) stations in Avaya Communication Manager. The SIP
telephones at the branch site use the IP network gateway as the default gateway. The MP-118 only
supports the operation of the analog endpoints and is not required for the SIP telephones. As a result,
CTM; Reviewed:
SPOC 11/29/2007
Solution & Interoperability Test Lab Application Notes
©2006 Avaya Inc. All Rights Reserved.
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AudioCodesMP118.doc
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