Aastra OpenCom 510 User Manual page 120

Voice over IP (VoIP)
which enables the SIP provider to provide calls to the telephone network.A SIP
connection can also accept incoming calls from the telephone network.
The same voice transmission techniques as those explained in Fundamentals
starting on page 110 are used for SIP telephony. SIP telephony has the following
distinctive features:
■
Subscribers are identified through an e-mail-like "SIP ID" such as
12345@domain.net or name@sip-provider.com.
■
SIP transmits dialling numbers always in a single data package (block dialling).
Dialling can therefore be concluded with the hash key
minal, or the end of the number will be indicated by a time-out. The value for
this time-out can be defined for each SIP provider separately.
■
You must log on ("Login") to the SIP registrar before you can use SIP telephony.
Use the OpenCom 510 to manage important information for the registration
(user name and password) of one or more SIP accounts. It is possible to make
several calls simultaneously using a single SIP account.
■
A SIP connection causes constant Internet data traffic, so do not use SIP with
Internet access which is paid for according to the time used.
■
RTP call data is also exchanged directly between terminals for SIP telephony, so
different codecs can be used for sending and for receiving. It is also possible to
change codecs dynamically during a call. You should use every codec available
in the VoIP profile at least once, because this will enable you to establish con-
nections with as many SIP subscribers as possible.
■
Fairly large packet lengths are quite normal on the Internet. They compensate
for the longer packet propagation delay.
■
A bidirectional RTP data stream with a dynamically-assigned UDP port number
is used to set up calls between subscribers. For this reason, incoming RTP calls
often fail to get past the Firewall or NAT configuration of the Internet gateway
product used. Do not use OpenCom 510 as an Internet gateway if the product
used is to be compatible with SIP telephony. These products provide a "Full
Cone NAT" setting for this application.
■
To enable the use of multiple devices on a single Internet connection, the IP
addresses used in a LAN (often 192.168.x.x) are translated to a valid IP address
using address translation (NAT - Network Address Translation), but no status
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SIP Telephony
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on the system ter-