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mise Delay" and "Maximise Throughput" for IP packets marked
with this value.
Call Set-up
Various IP-based protocols are used for system telephony via the
Internet protocol ("IP") (see also Start Procedure starting on
page 131). Multiple TCP connections are made between an IP tel-
ephone 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.
Channels on a Media Gateway (MGW) 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 connections 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 are only received by the other party if the
uncompressed G.711 codec is used. Alternatively, DTMF dial
tones can be transferred using the Internet standard
RFC 2833/4733. For this, the "DTMF (RFC4733)" codec needs
to be activated for the codec configuration on the Telephony:
Extended: VoIP profile page.
Analogue modems can not be used
Fundamentals
Voice over IP (VoIP)
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