Aastra OpenCom 510 User Manual page 123

Voice over IP (VoIP)
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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.
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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.
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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.
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Fairly large packet lengths are quite normal on the Internet. They compensate
for the longer packet propagation delay.
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A bidirectional RTP data stream with a dynamically-assigned UDP port number
is used to set up calls between subscribers. For this reason, incomng 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.
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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
information is available for NAT on an incoming RTP connection.To avoid this
problem, the IP address of a workplace computer or telephone visible on the
Internet is determined using a STUN server (STUN: Simple Traversal of UDP over
NAT). You can ask your SIP provider for the STUN server.'s IP address and port
number If you don't need a STUN server, leave the SIP Provider field empty.
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For direct SIP telephony using OpenCom 510, only SIP IDs consisting of
numbers for identifying subscribers registered with the SIP provider specified
can be addressed
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For each SIP account you can create just one bundle. You can specify this
bundle in routes as a connection option. You can use a network provider rule
to specify the routing of numbers within a specific range to use SIP telephony
as a preference (see also PBX Networking, under Configuration starting on
page 146).
SIP Telephony
121