Voice over IP (VoIP) Fundamentals
“Minimise Delay” and “Maximise Throughput” for IP packets marked with this value.
10.2.5 Call set-up
Various IP-based protocols are used for system telephony via the Internet protocol
(“IP”) (see also Start Procedure starting on page 125). Multiple TCP connections are
made between an IP telephone and OpenCom 510 for the telephone’s start procedure, registration and signalling.
Call data is directly exchanged between IP telephones using the RTP (“Realtime
Transport Protocol”) protocol.
Channels on a
Media Gateway (MGW) are allocated for making a telephone con-
nection 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.
10.2.6 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
■ 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
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