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SIP Telephony
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Voice over IP (VoIP)
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SIP transmits dialling numbers always in a single data package
(block dialling). Dialling can therefore be concluded with the
hash key
#
on the system terminal, or the end of the number
will be indicated by a time-out. The value for this time-out can
be defined for each SIP provider separately.
Tip:
The Forum 523/524 communications system can have a clip-
board for keeping track of the most recently dialled call numbers,
optimising en-bloc dialling. To do so, activate the
Dial out cache
option for a SIP line or a SIP tie-line bundle.
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You must log on (“Login”) to the SIP registrar before you can
use SIP telephony. Use the Forum 523/524 to manage
important information for the registration (user name and
password) of one or more SIP accounts. It is possible to make
several calls simultaneously using a single SIP account.
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A SIP connection causes constant Internet data traffic, so do
not use SIP with Internet access which is paid for according to
the time used.
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RTP call data is also exchanged directly between terminals for
SIP telephony, so different codecs can be used for sending and
for receiving. It is also possible to change codecs dynamically
during a call. You should use every codec available in the VoIP
profile at least once, because this will enable you to establish
connections with as many SIP subscribers as possible.
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Fairly large packet lengths are quite normal on the Internet.
They compensate for the longer packet propagation delay.
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A bidirectional RTP data stream with a dynamically-assigned
UDP port number is used to set up calls between subscribers.
For this reason, incoming RTP calls often fail to get past the
Firewall or NAT configuration of the Internet gateway product
used. If you do not use the Forum 523/524 as the Internet
gateway, the product should be compatible with SIP
telephony. These products provide a “Full Cone NAT” setting
for this application.
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To enable the use of multiple devices on a single Internet
connection, the IP addresses used in a LAN (often
192.168.x.x) are translated to a valid IP address using address
translation (NAT - Network Address Translation), but no status
information is available for NAT on an incoming RTP
Содержание Forum 523
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