SIP Telephony
102
Voice over IP (VoIP)
users pay to use and which enables the SIP provider to provide calls to the
telephone network. A SIP connection can also accept incoming calls from the
telephone network.
The same voice transmission techniques as those explained in
starting on page 92 are used for SIP telephony. SIP telephony has the
following distinctive features:
●
Subscribers are identified through an e-mail-like “SIP ID” such as
[email protected] or [email protected].
●
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.
●
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.
●
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.
●
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.
●
Fairly large packet lengths are quite normal on the Internet. They
compensate for the longer packet propagation delay.
●
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
Summary of Contents for forum 523
Page 1: ...Forum 523 524 Mounting and Commissioning Manual ...
Page 10: ...8 ...
