T-Comfort Comfort Pro S, Руководство по монтажу и пусконаладке

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SIP Telephony Voice over IP (VoIP)
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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 connections 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, incoming RTP calls often fail to get past the Firewall or NAT
configuration of the Internet gateway product used. If you do not use the Comfort Pro S as the Internet
gateway, the product should 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 workstation 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 Comfort Pro S, only SIP IDs consisting of numbers for identifying
subscribers registered with the SIP provider specified can be addressed
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You can integrate an external SIP connection in the Telephony : Trunks : Route menu into the route
configuration. 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 124).
You can configure SIP connections in the Configurator on the pages Telephony : Trunks : SIP provider and
Telephony : Trunks : SIP trunks . Enter the technical attributes of a specific SIP provider, such as the IP
addresses for the registrar and the STUN server under SIP provider . Under SIP trunks enter the information
for an existing SIP account, such as the user name, password, assigned call number and the maximum
number of simultaneous calls possible.