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SIP_STUN_KEEP_ALIVE_TIME defines in
seconds how often keep-alives are
sent in order to keep NAT bindings.
If SIP_STUN_BINDTIME_DETERMINE is
set, SIP_STUN_KEEP_ALIVE_TIME will be
overruled and keep alives will be
sent with a frequency of half of
determined bindtime
SIP_SIP_PORT
5060 (default) Local SIP port. Port used for first
UA instance. Succeeding UA’s will get
succeeding ports.
SIP_RTP_PORT
5004 (default) First port used for RTP traffic.
SIP_RTP_PORT_RANGE
20 (default)
Number of available RTP ports
SIP_RTP_PRIORITY
0 (default)
Priority of RTP traffic based on the
IP layer ToS byte. See RFC 1349 for
details.
Notice “cost bit” is not supported
Bit 7..5 defines precedence.
Bit 4..2 defines Type of Service.
Bit 1..0 are ignored.
Setting all three of bit 4..2 will be
ignored
SIP_SIP_PRIORITY
0 (default)
Priority of SIP call control
signalling traffic based on the IP
layer ToS byte. See RFC 1349 for
details.
Notice “cost bit” is not supported
Bit 7..5 defines precedence.
Bit 4..2 defines Type of Service.
Bit 1..0 are ignored.
Setting all three of bit 4..2 will be
ignored
Configuration file example:
%SIP_STUN_ENABLE%:0
%SIP_RPORT_ENABLE%:1
%NETWORK_STUN_SERVER%:"stun01.sipphone.com"
%SIP_STUN_BINDTIME_GUARD%:80
%SIP_STUN_BINDTIME_DETERMINE%:0
%SIP_STUN_KEEP_ALIVE_TIME%:90
%SIP_SIP_PORT%:5060
%SIP_RTP_PORT%:5004
%SIP_RTP_PORT_RANGE%:20
%SIP_RTP_PRIORITY%:0
%SIP_SIP_PRIORITY%:0
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