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Voice over IP (VoIP) SIP Telephony
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7.2.5 Call Set-up
Various IP-based protocols are used for system telephony via the Internet protocol
(“IP”) (see also Start Procedure starting on page 74). Multiple TCP connections are
made between an IP telephone and Aastra 800 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 from a Media Gateway (MGW) are allocated for making a telephone con-
nection with a SIP terminal or for dial tones. The MGW is a software function of the
Aastra 800 communications system. If the help of an MGW channel is necessary, IP
voice data are exchanged between the IP telephone and the PC which runs the
Aastra 800 communications system.
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.
7.3 SIP Telephony
The SIP Internet (Session Initiation Protocol) protocol provides you with a low-cost,
standardised option for telephoning via IP-based networks. Aastra 800 enables
you to use external SIP telephone connections (“SIP trunk lines”). Furthermore,
internal SIP subscribers, SIP telephones or SIP telephony software are also sup-
ported.
7.3.1 External SIP Connections
The Telephony: Trunks: Route menu gives you the option to configure a bundle
overflow, which automatically occupies a second line in case of a breakdown or
over-occupancy of the SIP connection. You can also set up your system to route
certain types of calls, such as international calls, to an SIP connection.
You will also need a fast Internet connection such as DSL for SIP telephony.
You will also usually need the services of a SIP provider. A SIP provider operates a
special server (the SIP Registrar) to handle connections. The SIP provider also