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VoIP SIP
VoIP SIP Servers, Users & the Forwarding Data-
iMG/RG Software Reference Manual (Voice Service)
6-22
Detaching a user from a port by means of the VOIP SIP USER REMOVE command, or, by deletion of the port
itself will result in a SIP de-registration transaction with the location server (assuming the user is registered
with the location server).
6.2.2.3 Forwarding database (FDB)
The forwarding database is a component of the iMG that is used to redirect calls to a different destination
address based on the called party number.
The signalling end-point layer uses the Forwarding DataBase every time the called end-point cannot be found
among the local users. It is used both for incoming calls from the VoIP network or for outgoing calls generated
locally and directed to a remote end-point.
The forwarding database may contain up to 100 entries (including users).
Forwarding entries are defined by the VOIP SIP FDB CREATE command.
Each FDB entry is uniquely identified by a name and defines the conditions that calls must satisfy in order to be
routed to the end point specified by FDB entry parameters.
When the signalling end-point layer receives a call it retrieves the called end-point address (called number).
Typically the called number is defined in the call signalling messages received from the network (in the
SIP
To
header).
If the call is locally originated the called number address is equal the dialled number (unless the ana-
logue/digital endpoint has the dialmask set to a value different from 0).
The Called end-point address is searched for among the local user addresses to establish whether the called
party is a user on the local system.
If the called end-point matches the address of a local user, the access port(s) associated with the called user
start ringing (if the port(s) are available).
If the called number cannot be found among the local users, the forwarding database is scanned to look for
entries matching the called number.
Note that the forwarding algorithm acts differently depending on whether the call is locally originated, or, is an
incoming call:
6.2.2.3.1 Locally originated calls
If a match is found, the INVITE message is routed to the IP address defined in the CONTACT field of the
matched FDB entry.
The called user domain will be set to the DOMAIN value (optional) or to the CONTACT
value (if no DOMAIN is specified) defined by the DOMAIN and CONTACT fields in the FDB entry respec-
tively.
If the FDB entry has defined the FWADDRESS field, the called number is changed from the dialed number to
the number defined in the FDB entry FWADDRESS field. In this way it's possible to dial short numbers that will
be replaced by full-qualified numbers in outgoing calls.
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VoIP SIP Servers, Users & the Forwarding Database
VoIP SIP
6-23
iMG/RG Software Reference Manual (Voice Service)
If no match is found in the forwarding database, the INVITE message is routed to the first available proxy server
(starting with the Master proxy server if defined) using the calling user domain as called endpoint domain.
6.2.2.3.2 Incoming calls
If a match is found, a MOVED TEMPORARY SIP message is sent back to the call originator reporting the contact
address defined by the CONTACT field in the matched FDB entry.
If the FDB entry defines the FWADDRESS field, the called number is changed from the dialed number to the
number defined in the FDB entry FWADDRESS field.
If no match is found in the forwarding database, the call is rejected.
6.2.2.3.3 Address and digit-map
The address field specified in FDB entries can be defined using digit map expressions.
Digit map expressions are used to increase system flexibility when defining forwarding rules that must mach
multiple addresses (digit maps are used also in the VoIP access port module).
A digit map is defined either by a case insensitive ‘string’, or by a list of strings. Each string in the list is an alter-
native numbering scheme, specified either as a set of digits or as an expression to which the called address is
compared by the
signalling
end-point layer to find the shortest possible match. The following constructs can be
used in each digit map:
Digit
A digit from '0' to '9'
Wildcard
The symbol ‘x’ that matches any digit (‘0’ to ‘9’).
Range
One or more digit symbols enclosed between square brackets (‘[‘ and ‘]’).
Subrange
Two digits separated by hyphen (‘-’) that matches any digit between and including the two. The subrange
construct can only be used inside a range construct, i.e. between ‘[‘ and ‘]’.
Position
A period (‘.’) that matches an arbitrary number, including zero, of occurrences of the preceding construct.
Digit map expressions are typically used when managing locally originated calls.
Using digit map expressions in this situation, it is possible to define a generic rule in such a way that all calls are
routed to a specific contact (e.g. the proxy server) which will then perform call routing.
Digit map expressions are also useful for designing small networks without need to make use of any location
servers, proxy servers or gatekeepers.
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VoIP SIP
VoIP SIP Embedded Proxy Server
iMG/RG Software Reference Manual (Voice Service)
6-24
6.2.3
VoIP SIP Embedded Proxy Server
All gateway models with the exception of RG600E and RG6x6E variants include support for the embedded SIP
proxy server. See table 1 (RG/iMG Models) for further details.
Refer to section
6.2.7
for the Embedded Proxy Server (EPS) CLI commands.
Also, note the following rules and guidelines for SIP:
The maximum number of sip fdb users is 128, except for the iMG616E (64).
The media port limit depends on the cpu type, and so the following number of ports are available:
iMG616E (Helium-210) - up to 48
iMG634A/B, iMG634WA/B, iBG910A/B (Argon-4x2) - up to 48
iBG915FX, iMG6x6MOD, iMG7x6MOD (Helium-520) - up to 128
iMG634A/B-R2, iMG634WA/B-R2, iMG616W (Solos) - up to 128
The default value is always 32.
Note:
Do not use the SECURITY ADD ALG command with the SIP option when configuring EPS, as this will
cause issues with managing NAT sessions.
Note:
When configuring EPS, note that EPS allows a maximum of three calls per line, although some IP-
phones can support more than three.
6.2.4
VoIP SIP command reference
This section describes the commands available on the iMG to configure and manage the SIP protocol-signalling
module.
6.2.4.1 VoIP SIP protocol CLI commands
The table below lists the VOIP SIP protocol commands provided by the CLI:
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VoIP SIP command reference
VoIP SIP
6-25
iMG/RG Software Reference Manual (Voice Service)
TABLE 6-4
VoIP SIP Protocol CLI Commands
6.2.4.1.1 VOIP SIP PROTOCOL DISABLE
Commands
Fiber
A
Fiber
B
Fiber
C
Fiber
D
Fiber
E
Modular
ADSL
A
ADSL
B
ADSL
C
VOIP SIP PROTOCOL DISABLE
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL ENABLE
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL RESTART
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET AUTHENTICATION
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET CONTACT-ON-1XX-RESPONSE
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET DEFAULTPORT
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET EXTENSION
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET INFO
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET INTERNAL-CALL-ROUTING
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET INVITETIMEOUT
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET KEEP-ALIVE
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET NAT
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET NETINTERFACE
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET PATH-HEADER
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET REGISTRATION-RETRY-TIME
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET REGISTRATION-RING-SPLASH
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET REMOTE-PARTY-ID-REPLACEMENT-ON-CFWD
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET ROUNDTRIPTIME
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET SERVER-REDUNDANCY
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET SERVER-SWITCHING
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET SESSIONEXPIRE
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET SUBSCRIBE-EVENT-MESSAGE-SUMMARY
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET UNRESERVED-CHAR-EXTENSION
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SET URIHOST
X
X
X
X
X
X
X
X
X
VOIP SIP PROTOCOL SHOW
X
X
X
X
X
X
X
X
X
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VoIP SIP
VoIP SIP command reference
iMG/RG Software Reference Manual (Voice Service)
6-26
Syntax
VOIP SIP PROTOCOL DISABLE
Description
This command stops the VoIP SIP signalling protocol and releases all the resources asso-
ciated to it:
Any analogue or digital port defined in the system is removed
Any user defined in the system is deleted
Any forwarding entry in the FDB is deleted
Any sip server reference (location and proxy) is removed
To simply restart the SIP module, use the VOIP SIP PROTOCOL RESTART command. It
doesn't remove any resources defined under the VoIP main module.
To enable the SIP module, use the VOIP SIP PROTOCOL ENABLE command.
Example
--> voip sip protocol disable
See also
VOIP SIP PROTOCOL RESTART
VOIP SIP PROTOCOL ENABLE
6.2.4.1.2 VOIP SIP PROTOCOL ENABLE
Syntax
VOIP SIP PROTOCOL ENABLE
Description
This command turns on the SIP
signalling
module.
To bind the SIP module to a specific IP interface use the VOIP SIP PROTOCOL SET
INTERFACE command.
Note:
Binding the SIP module to a specific IP interface defines the value of the source IP address for signalling
and voice packets. SIP URLs with local reference offer the hostname and the IP address belonging the
provisioned interface.
Note:
The SIP module MUST be enabled in order to create/set analog/digital ports, users, call forwarding rules
and SIP servers.
Example
--> voip sip protocol enable
See also
VOIP SIP PROTOCOL SHOW
VOIP SIP PROTOCOL DISABLE
6.2.4.1.3 VOIP SIP PROTOCOL RESTART
Syntax
VOIP SIP PROTOCOL RESTART
Description
This command restarts the VoIP SIP
signalling
protocol module.
Any pending and active calls are released.

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