Voice over IP (VoIP)
115
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 con-
nections 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 acti-
vated for the codec configuration on the Telephony: Extended: VoIP profile
page.
• 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.
VoIP Profiles for SIP
While a call is being established and also during an ongoing connection, SIP terminals
can negotiate amongst one another which codecs are to be used. SIP signalling is
always being conducted via the OpenCom 100 communications system. After suc-
cessfully agreeing upon a codec, the resulting RTP connection can be established
between the terminals directly. The OpenCom 100 communications system can also
serve as a proxy for the media data stream (“RTP Proxy function”) if a direct RTP data
connection is not feasible between the terminals.
Negotiating the codecs to be used takes place as follows:
To Next Page
Loading...
Need help?
General
