Lancom Systems DSL/I-10+ LS61133 Benutzerhandbuch

LANCOM DSL/I-10+

VoIP (optional)

Extended VoIP support (only in combination with LANCOM VoIP options)

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

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

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.

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

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

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

Number of local subscribers

4 SIP (VoIP Basic Option), 32 SIP (VoIP Advanced Option)

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. ISDN supplementary services CLIP, CLIR, 
en- block dial and individual dialing with adjustable wait- time until completion. ISDN- UDI calls with G.722 (not with VoIP Basic 
Option). 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).

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 (not with VoIP Basic Option), G.729 Annex 
A (not with VoIP Basic Option)

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

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

Router

IP, IPX and NetBIOS/IP multi- protocol router

HTTP

HTTP and HTTPS server for configuration by web interface

DNS

DNS client, DNS server, DNS relay, DNS proxy and dynamic DNS client

DHCP

DHCP client, DHCP relay and DHCP server with autodetection

NetBIOS

NetBIOS/IP proxy

NTP

NTP client and SNTP server, automatic adjustment for daylight- saving time

Policy- based routing

Policy- based routing based on routing tags. Based on firewall rules, certain data types are marked for specific routing, e.g. to 
particular remote sites or lines.

Dynamic routing

Dynamic routing with RIPv2. Learning and propagating routes; separate settings for LAN and WAN

IP

ARP, proxy ARP, BOOTP, LANCAPI, DHCP, DNS, HTTP, HTTPS, IP, ICMP, NTP/SNTP, NetBIOS, PPPoE (server), RADIUS, RIP- 1, RIP-
2, RTP, SIP, SNMP, TCP, TFTP, UDP, VRRP

IPX

RIP, SAP, IPX and SPX watchdogs, NetBIOS watchdogs

Ethernet

PPPoE, Multi- PPPoE, ML- PPP, PPTP (PAC or PNS) and plain Ethernet (with or without DHCP), RIP- 1, RIP- 2

ISDN

1TR6, DSS1 (Euro- ISDN), PPP, X75, HDLC, ML- PPP, V.110/GSM/HSCSD, CAPI 2.0 via LANCAPI, Stac data compression