Voice over IP (VoIP)
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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.
• 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. If you do not use the OpenCom 100 as the Internet gateway, the product
should 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 infor-
mation 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.
• For direct SIP telephony using OpenCom 100, only SIP IDs consisting of numbers
for identifying subscribers registered with the SIP provider specified can be
addressed
• 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 pref-
erence (see also PBX Networking, under Configuration starting on page 166).
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.
For extension-capable SIP-DDI lines, you can, as needed, make additional settings on
the Telephony: Trunks: SIP Provider page. Assigning extension call numbers and
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