BELGACOM Forum 526, Руководство по монтажу и пусконаладке

Страница: 109 / 232

SIP Telephony 107
Voice over IP (VoIP)
z
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.
z
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.
z
Fairly large packet lengths are quite normal on the Internet.
They compensate for the longer packet propagation delay.
z
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 Forum 525/526 as the Internet
gateway, the product should be compatible with SIP
telephony. These products provide a “Full Cone NAT” setting
for this application.
z
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.
z
For direct SIP telephony using Forum 525/526, only SIP IDs
consisting of numbers for identifying subscribers registered
with the SIP provider specified can be addressed
z
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