AudioCodes Mediant 2000 User Manual

Version 6.6 245 Mediant 2000
User's Manual 20. IP-to-IP Routing Application
20 IP-to-IP Routing Application
The device's IP-to-IP application supports IP-to-IP VoIP call routing (or SIP Trunking). The
IP-to-IP call routing application enables enterprises to seamlessly connect their IP-based
PBX (IP-PBX) to SIP trunks, typically provided by Internet Telephony Service Providers
(ITSP). The device enables the enterprise to communicate with the PSTN network (local
and overseas) through the ITSP, which interfaces directly with the PSTN. Therefore, the
IP-to-IP application enables enterprises to replace the bundles of physical PSTN wires with
SIP trunks provided by ITSPs and use VoIP to communicate within and outside the
enterprise network using its standard Internet connection. At the same time, the device can
also provide an interface with the traditional PSTN network, enabling PSTN fallback in case
of IP connection failure with the ITSPs.
The device also supports multiple SIP Trunking. This can be useful in scenarios where if a
connection to one ITSP fails, the call can immediately be transferred to another ITSP. In
addition, 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).
In addition to providing VoIP communication within the enterprise's LAN, the device
enables the enterprise to communicate outside of the corporate LAN using SIP Trunking.
This includes remote (roaming) IP-PBX users, for example, employees using their laptops
to communicate with one another from anywhere in the world such as at airports.
The IP-to-IP application can be implemented by enterprises in the following example
scenarios:

VoIP between an enterprise's headquarters and remote branch offices

VoIP between an enterprise and the PSTN through an ITSP
The IP-to-IP call routing capability is feature-rich, allowing interoperability with different
ITSPs:

Easy and smooth integration with multiple ITSP SIP trunks.

Supports SIP registration and authentication with ITSP servers (on behalf of the
enterprise's IP telephony system) even if the enterprise's IP telephony system does no
support registration and authentication.

Supports SIP-over-UDP, SIP-over-TCP, and SIP-over-TLS transport protocols, one of
which is generally required by the ITSP.

Provides alternative routing to different destinations (to another ITSP or the PSTN)
when the connection with an ITSP network is down.

Provides fallback to the legacy PSTN telephone network upon Internet connection
failure.

Provides Transcoding from G.711 to G.729 coder with the ITSP for bandwidth
reduction.

Supports SRTP, providing voice traffic security toward the ITSP.

IP-to-IP routing can be used in combination with the regular Gateway application. For
example, an incoming IP call can be sent to an E1/T1 span or it can be forwarded to
an IP destination.
Therefore, the device provides the ideal interface between the enterprise IP-PBX and the
ITSP SIP trunk.
The device's IP-to-IP application handles and terminates SIP methods and responses at
each leg independently:

Initiating-dialog INVITE: terminated at one leg and initiated on the other leg,
180\182\183\200\4xx uses the same logic and same limitations, in some cases the
result may be a different response code.

OPTIONS: terminated at each leg independently.

INFO: only specific INFO’s (such as DTMF) are handled; other types are omitted.

AudioCodes Mediant 2000 Specifications

General

Brand	AudioCodes
Model	Mediant 2000
Category	Gateway
Language	English