Routing Applications; Ip-To-Ip Routing Application - AudioCodes Mediant 1000 User Manual

9.2

Routing Applications

9.2.1

IP-to-IP Routing Application

The device's 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 an Internet Telephony Service Provider (ITSP). By
implementing the device, enterprises can then communicate with PSTN networks (local and
overseas) through ITSP's, which interface 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 ITSP's 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 ITSP's.
In addition, the device 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).
Therefore, in addition to providing VoIP communication within an enterprise's LAN, the
device allows 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 via an ITSP.
The IP-to-IP call routing capability is feature-rich, allowing interoperability with different
ITSP's or service providers:

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 enterprises' IP-PBX's and ITSP
SIP trunks. To facilitate the understanding of the IP-to-IP feature, this section provides a
configuration example.
SIP User's Manual 492
Mediant 600 & Mediant 1000
Document #: LTRT-83306