Fundamentals
96
Voice over IP (VoIP)
Call Set-up
Various IP-based protocols are used for system telephony via the Internet
protocol (“IP”) (see also
starting on page 114). Multiple TCP
connections are made between an IP telephone and Forum 523/524 for the
telephone’s start procedure, registration and signalling.
All voice data are directly exchanged between IP telephones using the RTP
(“Realtime Transport Protocol”) protocol.
are allocated for making a telephone
connection with an ordinary terminal or for dial tones. The MGW converts IP
voice data into PCM data streams used with conventional telephony and vice
versa. For this, IP voice data are exchanged between the IP telephone and the
gateway.
Tip:
Switching between voice data channels may cause a slight delay
in some circumstances. For example: when accepting a call on an
IP telephone, headset users should wait about one second before
answering.
Useful Services
The type of data compression used for VoIP prevents these types of connec-
tions from using certain services. Take these notes into account especially if
you want to use connections made via Q.SIG-IP or SIP:
●
ISDN data services can not be used
●
Faxes can only be sent using the uncompressed G.711 codec
●
DTMF dial tones can be transmitted using the internet standard RFC 2833/
4733 or by means of SIP-INFO. In the codec configuration section, under
Telephony: Extended: VoIP profile
the “DTMF (RFC 4733)” codec hast to
be activated. As an alternative, DTMF dial tones can also be transferred
“Inband”. These DTMF dial tones are only received by the other party if the
uncompressed G.711 codec is used.
●
Analogue modems can not be used
Tip:
Configure the actual usage for the a/b ports, e.g. set them to
Fax
or
Data (analogue)
. Connections from and to these a/b ports
will then be made using uncompressed or ISDN connections
where possible.
Summary of Contents for forum 523
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