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112
Fundamentals
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 con-
nection, SIP terminals can negotiate amongst one another which
codecs are to be used. SIP signalling is always being conducted
via the Forum 523/524 communications system. After success-
fully agreeing upon a codec, the resulting RTP connection can be
established between the terminals directly. The Forum 523/524
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:
1. One of the SIP terminals sends of list of supported codecs.
2. The second SIP terminal deletes the codecs from the list which
are not supported and sends the shortened list back to the first
SIP terminal.
The exchanged codec lists can be filtered by the Forum 523/524
communications system, e.g. to limit bandwidth based on the
VoIP profile activated for the connection or for the terminal.
Please note that the negotiation regarding codecs supported also
takes place for connections between SIP terminals and other
terminals / subscribers. For example, the Forum 523/524 commu-
nications system uses a pre-defined codec list for a connection
between an SIP terminal and a TDM port:
G.711a
IP system phones or DECT over IP base stations participate in
codec negotiation with the VoIP profile configured respectively. If
no common audio codec is determined during negotiation, it will
not be possible to establish the call connection.
Voice over IP (VoIP)
G.729 (if possible)
RFC4733 DTMF
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