Lancom Systems 1722 LS61350 Benutzerhandbuch
Produktcode
LS61350
LANCOM 1722 VoIP
NAT- Traversal
NAT- Traversal (NAT- T) support for VPN over routes without VPN passthrough
IPCOMP
VPN data compression based on LZS or Deflate compression 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 dial- in for remote sites via connection template
the ICMP or UDP protocol in encrypted form. Dynamic dial- in for remote sites via connection template
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
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.
names are translated by Internet DNS servers.
VPN throughput (max.)*
1364- byte packet size
24 Mbps
265- byte packet size
6 Mbps
Notice
* all VPN figures with AES encryption and active VPN hardware acceleration
Firewall throughput (max.)
1470- byte packet size
72 Mbps
256- byte packet size
9 Mbps
VoIP
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/remove line- related prefixes or switchboard numbers
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/remove line- related prefixes or switchboard numbers
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 providers and SIP PBXs with automatic login of SIP users at SIP providers/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, service location (SRV) support
as telephone lines for shared use. Connection to up to four upstream SIP PBXs including line backup. SIP connections from/to
internal subscribers, SIP providers and SIP PBXs with automatic login of SIP users at SIP providers/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, service location (SRV) support
SIP gateway
Transparent conversion of ISDN telephone calls to SIP calls, and vice versa. Local ISDN subscribers register as local SIP users,
and local ISDN subscribers automatically register as SIP users at upstream SIP PBXs/with SIP providers. Number translation
between internal numbers and MSN/DDI (including telephone number blocks) or external numbers, plus automatic adaptation
of calling numbers and called numbers at the transition.
and local ISDN subscribers automatically register as SIP users at upstream SIP PBXs/with SIP providers. Number translation
between internal numbers and MSN/DDI (including telephone number blocks) or external numbers, plus automatic adaptation
of calling numbers and called numbers at the transition.
SIP trunk
Outgoing call switching and incoming call reception based on extension numbers to/from SIP PBXs/SIP providers (requires
support of the SIP- DDI functions compliant with ITU- T Q.1912.5 at the central exchange) with just a single user account to
register the switchboard number; mapping of entire SIP telephone number blocks
support of the SIP- DDI functions compliant with ITU- T Q.1912.5 at the central exchange) with just a single user account to
register the switchboard number; mapping of entire SIP telephone number blocks
SIP link
Outgoing call switching and incoming call reception of any numbers to/from SIP PBXs/SIP providers (requires support of this
function at the central exchange) with just a single user account to register the switchboard number; mapping of entire SIP
telephone number blocks
function at the central exchange) with just a single user account to register the switchboard number; mapping of entire SIP
telephone number blocks
SIP remote gateway
Local break- in/out of calls with any telephone number to/from upstream VoIP PBXs/SIP providers with telephone number
mapping; independent of local users
mapping; independent of local users
Switching and call routing functions
Switching between local SIP subscribers and upstream SIP PBX or SIP subscribers and ISDN/analog subscribers (depending on
connection types) initiated by SIP client
connection types) initiated by SIP client
Number of local subscribers
32 SIP, ISDN unlimited (max. 40 mapping entries)
Number of simultaneous connections
2 - 16 depending on code conversion, echo canceling and load
Signaling
VoIP: SIPv2, ISDN: DSS1 (Euro- ISDN), point- to- point/point- to- multipoint; 1TR6 (only at an external ISDN connector in TE mode)
Media protocols
RTP
ISDN features
Operation direct at ISDN exchange lines or at ISDN extension lines of existing PBXs. Provision of exchange lines or extension
lines. ISDN supplementary services CLIP, CLIR, en- block dial and individual dialing with adjustable wait- time until completion.
Transparent pass- through of data services. ISDN- UDI calls with G.722. Pass- through of service identifiers (BC, HLC, LLC) for
ISDN- to- ISDN connections. PCM bit- transparent coupling. Support for keypad facilities. Advice of charge (AOC- D, AOC- E).
"DSS1 NT reverse" and "DSS1 NT point- to- point reverse" for ISDN clock synchronization with suitable PBXs. ISDN S0 buses can
be collected into hunting groups. Parallel operation of point- to- point and point- to- multipoint connections
lines. ISDN supplementary services CLIP, CLIR, en- block dial and individual dialing with adjustable wait- time until completion.
Transparent pass- through of data services. ISDN- UDI calls with G.722. Pass- through of service identifiers (BC, HLC, LLC) for
ISDN- to- ISDN connections. PCM bit- transparent coupling. Support for keypad facilities. Advice of charge (AOC- D, AOC- E).
"DSS1 NT reverse" and "DSS1 NT point- to- point reverse" for ISDN clock synchronization with suitable PBXs. ISDN S0 buses can
be collected into hunting groups. Parallel operation of point- to- point and point- to- multipoint connections
Audio properties
Echo canceling (G.168), automatic adaptive de- jitter buffer. Inband tone signaling compliant with EU standards and country-
specific. DTMF support compliant with RFC 2976 (SIP info), RFC 2833 (RTP payload type/outband). Transparent pass- through
for negotiated codecs. Interaction on codec negotiation between subscribers (filter, quality/bandwidth) Voice encoding with
G.711 μ- law/A- law (64 kbps), G.726 (16, 24, 32, 40 kbps), G.722 high- quality codec, G.729 Annex A
specific. DTMF support compliant with RFC 2976 (SIP info), RFC 2833 (RTP payload type/outband). Transparent pass- through
for negotiated codecs. Interaction on codec negotiation between subscribers (filter, quality/bandwidth) Voice encoding with
G.711 μ- law/A- law (64 kbps), G.726 (16, 24, 32, 40 kbps), G.722 high- quality codec, G.729 Annex A
Auto QoS
Automatic dynamic bandwidth reservation per SIP connection. Automatic selection of compression method depending upon
available bandwidth. Voice packet prioritization (CoS), DiffServ marking, traffic shaping (incoming/outgoing) and packet- size
management of non- prioritized connections compared to VoIP
available bandwidth. Voice packet prioritization (CoS), DiffServ marking, traffic shaping (incoming/outgoing) and packet- size
management of non- prioritized connections compared to VoIP
VoIP management
VoIP Setup Wizard in LANconfig; status display of subscribers, lines, and connections; logging of VoIP Call Manager events in
LANmonitor. SYSLOG and TRACE for voice connections
LANmonitor. SYSLOG and TRACE for voice connections
VPN