Datasheet

LANCOM 1823 VoIP
Algorithms 3DES (168 bit), AES (128, 192 or 256 bit), Blowfish (128 - 448 bit), RSA (128 or -448 bit) and CAST (128 bit); MD-5 or SHA-1
hashes
NAT traversal NAT traversal (NAT-T) support for VPN over routes without VPN passthrough
IPCOMP VPN data compression based on LZS or deflate compression algorithms for higher IPSec throughput
LANCOM Dynamic VPN Enables VPN connections from or to dynamic IP addresses. The IP address is communicated via ISDN B- or D-channel or with
the ICMP or UDP protocol in encrypted form
Dynamic DNS (dynDNS) Enables the registration of IP addresses with a dynDNS provider in the case that fixed IP addresses are not used for the VPN
connection
Specific DNS forwarding DNS forwarding according to DNS domain, e.g. internal names are translated by proprietary DSN servers in the VPN; external
names are translated by Internet DNS servers
VPN throughput (max.)*
1364 Byte paket size 24 Mbps
265 Byte paket size 6 Mbps
Hint * by AES encryption and active VPN hardware acceleration
Firewall throughput(max.)
1470 Byte paket size 72 Mbps
256 Byte paket size 9 Mbps
VoIP
SIP Proxy
Management of local SIP users with optional automatic registration/authentication. Mapping of public SIP provider accounts
as telephone lines for shared use. Connection to up to four upstream SIP PBXs including line backup. SIP connections from/to
internal subscribers, SIP provider and SIP PBXs with automatic registration of SIP users at SIP provider/upstream SIP PBXs.
Optional shared/individual password for authentication at an upstream SIP PBX. Automatic bandwidth management and
automatic configuration of the firewall for SIP connections. Default DNS entry for the local SIP domains, support of service
location (SRV)
SIP gateway
Transparent transition of analog or ISDN calls to SIP and vice versa. Local ISDN and analog subscribers register as local SIP
users, and local ISDN or analog subscribers automatically register as SIP users at upstream SIP PBXs/SIP providers. Number
translation between internal numbers and MSN/DDI or external number and automatic adaptation of caller numbers and cal-
led numbers at the transition
Call router
Central switching of all incoming and outgoing calls. Number translation by mapping, numeral replacement and number
supplementation. Configuration of line and route selection, entry of multiple alternative routes (line backup). Routing based
on calling and called number, SIP domain and line. Manual routing by the user ("outside-line access codes"); routing with
line selection keys on telephones or telephone number prefixes; targeted routing for individual telephone numbers (e.g.
emergency calls via local ISDN); separate routes for internal, local, long-distance or international calls; blocking of telephone
numbers or blocks of telephone numbers; inclusion of local subscribers into the number range of an upstream SIP PBX;
internal standard telephone number for undeliverable calls; supplement/removal of line-related prefixes or trunk numbers
SIP trunking
Switching of outgoing calls and acceptance of incoming calls based on direct dial in numbers to/from SIP PBXs/SIP providers
(support of SIP DDI functionality according to ITU-T Q.1912.5 necessary at the upstream SIP exchange) with a single user
account to register the base number, mapping of whole blocks of SIP numbers
SIP link
Switching of outgoing calls and acceptance of incoming calls with any number to/from SIP PBXs/SIP providers (upstream SIP
exchange must support this feature) with a single account to register, mapping of whole blocks of SIP numbers
SIP remote gateway
Dial-in or dial-out from/to the local telephone network at a remote site with number translation made available for a central
SIP PBX/SIP provider
Switching functions Switching between local SIP subscribers and upstream SIP PBXs as well as SIP and ISDN or analog subscribers (depending on
the ports) when initiated by a SIP client
Number of local users 32 SIP, unlimited ISDN (max. 40 mapping entries)
Number of parallel connections 2 - 16 depending on code conversion, echo canceling and load
Signalling VoIP: SIPv2, ISDN: DSS1 (Euro ISDN), 1TR6 (only at an external ISDN connection in TE mode)
Media protocols RTP
ISDN modes Operation directly at ISDN exchange lines or ISDN extension lines of PBXs. Support of exchange oder extension lines. ISDN
facilities CLIP, CLIR, block dialing, overlap sending with configureable delay for number completion. Transparent pass-
through of data services. ISDN UDI calls with G.722. Signalisation of calltype (BC, HLC, LLC) for connections from ISDN to
ISDN. PCM bit-transparent connections. Support of keypad facilities. Transmission of advice of charge (AOC-D, AOC-E). 'DSS1
NT reverse' and 'DSS1 NT point to point reverse' for synchronisation with external ISDN timer of PBXs supporting this fea-
ture. Configuration of multiple ISDN S
0
busses to trunk lines. Simultaneous support for multiple subscriber lines and direct
dialing in lines