NEC Univerge SV8100 Features And Specifications Manual page 870

Issue 13.0
Auto start video when call is answered is not supported.
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Standard SIP video Codecs are not supported across CCIS.
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Standard SIP video Codecs are not supported across NetLink.
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Standard SIP terminal can not negotiate video Codec with SP310.
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SIP protocol (RFC3261) is used.
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SIP Station uses the PZ-( )IPLA/IPLB as a media gateway.
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Default UDP listen port for a SIP station is 5070.
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UNIVERGE SV8100 Station registration policy supports an authentication feature. Enabling
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this policy prevents the registered telephone from unexpected override.
UNIVERGE SV8100 supports HOLD and TRF feature on the basis of
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draft-ietf-sipping-service-examples-09.txt
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Section 2.5 (Transfer - Attended) of draft-ietf-sipping-service-examples-15.txt
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draft-ietf-sip-session-timer-10.txt
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When all VoIP DSP resources are busy, the SIP phone cannot preempt active calls to
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make a 911 call.
The UNIVERGE SV8100 CD-CP00-US is the registration server for the SIP stations. The
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configurable IP Address is located in Program 10-12-09 (SV8100 Network Setup – IP
Address).
T.38 (Fax) is not supported for 3rd Party SIP "IP Single Line Telephone (SIP)" station ports
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(Version 3100 or lower software).
T.38 (Fax) is supported for 3rd Party SIP "IP Single Line Telephone (SIP)" station ports
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(Version 4000 or higher software).
Program 15-03-03 must be set to 1 (Special) at the receiving terminal in order for T.38 to
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function.
With SV8100 Version 5000 (5.00 or higher) software and PZ-IPLB daughter board
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installed, half duplex connections are not supported. For troubleshooting purposes, a
managed switch capable of port mirroring is required to capture packet data from the
SV8100 IPLB Ethernet port.
When using 3rd party SIP stations, the SIP server name can not contain a parenthesis.
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When out of band DTMF is used (via RFC2833), the IPLA supports a payload of 96 ~ 126.
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IPLB supports an out of band DTMF payload of 96 ~ 127.
If a user on a standard SIP phone is talking to another station using Voice Announce (the
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receiving station has not pressed speaker or lifted the handset) and the SIP phone user
presses transfer or hold, the call is terminated. A standard SIP call cannot be placed on
hold or transferred until the other party answers.
2 - 836
UNIVERGE SV8100
draft.
IETF
IP Single Line Telephone (SIP)