AudioCodes Mediant 500 E-SBC, User Manual

Page: 444 / 920

User's Manual 444 Document #: LTRT-10437
Mediant 500 E-SBC
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Layer-3 topology hiding by modifying source IP address in the SIP IP header.
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SIP normalization: The device supports SIP normalization, whereby the SBC
application can overcome interoperability problems between SIP user agents. This is
achieved by the following:
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Manipulation of SIP URI user and host parts.
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Connection to ITSP SIP trunks on behalf of an IP-PBX - the device can register
and utilize user and password to authenticate for the IP-PBX.
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Survivability:
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Routing calls to alternative routes such as the PSTN.
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Routing calls between user agents in the local network using a dynamic database
(built according to registrations of SIP user agents).
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Routing:
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IP-to-IP routing translations of SIP, UDP, TCP, TLS (when extensive transcoding
is not required).
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Load balancing and redundancy of SIP servers.
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Routing according to Request-URI\Specific IP address\Proxy\FQDN.
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Alternative routing.
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Routing between different Layer-3 networks (e.g., LAN and WAN).
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Load balancing\redundancy of SIP servers.
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ITSP accounts.
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SIP URI user and host name manipulations.
27.1 SIP Network Definitions
The device's SBC application can implement multiple SIP signaling and RTP (media)
interfaces.
27.2 SIP Dialog Initiation Process
The device's SIP dialog initiation process concerns all incoming SIP dialog initiation
requests. This includes SIP methods such as INVITE, SUBSCRIBE, OPTIONS, REFER,
INFO, UNSOLICITED NOTIFY, MESSAGE, and REGISTER.
The SIP dialog initiation process consists of the following stages:
1. Determining source and destination URL: The SIP protocol has more than one
URL in a dialog-establishing request that may represent the source and destination
URLs. When handling an incoming request, the device uses specific SIP headers for
obtaining the source and destination URLs. Once these URLs are determined, their
user and host parts are used as input for the classification process, message
manipulation, and call routing.
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All SIP requests (e.g., INVITE) except REGISTER dialogs:
♦
Source URL: The source URL is obtained from the SIP header according to
the following logic:

The source URL is obtained from the From header.

If the From header contains the value 'Anonymous', the source URL is
obtained from the P-Preferred-Identity header.

If the P-Preferred-Identity header does not exist, the source URL is
obtained from the P-Asserted-Identity header.
♦
Destination URL: The destination URL is obtained from the Request-URI.