DS Series IP Guide

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IP FEATURES GUIDE

DIGITAL TELEPHONE EXCHANGES


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DS SERIES

DIGITAL TELEPHONE EXCHANGES

IP FEATURESGUIDE

NOVEMBER – 2010


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DS Series IP Features Guide

II

DS SERIES IP FEATURES GUIDE – REV. AAA – 01.12.2010KAREL reserves the right to make modifications in product features mentioned in this document for development

and improvement purposes, without prior notice. Individual products may possess characteristics different fromthose that have been mentioned in this document, due to their differences in software and hardware versions .

VERSION TABLECPU SOFTWARE VERSIONS DATE/VERSION OF GUIDE

BAD AAA/01.12.2010


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PREFACE

This guide aims to cover all IP related programming in DS Series Systems.

First section is dedicated to basic IP parameters and their controls on a network. They are included inthis guide to be a general guidance to the technical staff who will be putting IP extensions, lines or

gateways into service.Next sections are totally specific to Karel DS Series Systems and Karel’s own IP programmingsolutions are given. Necessary programming for IP extensions, IP lines and IP gateways are given inseparate sections. The first programming section is dedicated to EX200 MGW Media Gateway card,which is the backbone of communications as for IP extensions and IP lines.

Last sections describe tools that can used to monitor IP communications in Karel systems andnecessary precautions for secure IP communication.

We wish you a successful programming session.

KAREL Electronics


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CONTENTSI. INTRODUCTION ................................... ........................................ ......................................... ............. 1II. CONTROLS BEFORE SYSTEM PROGRAMMING ..................................... ..................................... 2II.1. CONTROL STEPS FOR BASIC COMMUNICATION ............................................. ........................ 2

II.2. CONTROL STEPS FOR VOICE QUALITY................. .......................................... .......................... 5II.3. CONTROL STEPS FOR NETWORK DEVICES ............................................. ................................ 6II.4. BANDWIDTH usage by Karel modules........................................... .......................................... ...... 7II.4. THE LIST OF KAREL MODULES ........................................ ........................................ .................. 8III. VoIP GATEWAY ........................................ ........................................ ........................................ ....... 9III.1. VOIP GATEWAY CARD TECHNICAL SPECIFICATIONS ............................................. ............... 9III.2. FUNCTIONS OF A VoIP GATEWAY CARD .................................... ........................................... . 10III.3. PROGRAMMING ON THE SYSTEM......................................... ............................................. ..... 10III.4. PROGRAMMING on vo ı p gateway card ........................................... ........................................... 11III.4.1. NETWORK SETTINGS......... ............................................ ............................................. ........... 11III.4.2. ADDRESS TABLE ..................................... .......................................... ..................................... 11III.4.3. CAPABILITY TABLE ......................................... .......................................... .............................. 11III.4.4. PARAMETERS .................................... ........................................... .......................................... 12

III.4.5. REGISTRATION PROXY............................ ............................................ .................................. 12III.4.6. WRITE TO EEPROM....................................... ........................................... .............................. 12III.4.7. MORE OPTIONS ON KNEE ............................................ ............................................. ............ 12IV. MGW CARD PROGRAMMING ...................................... ......................................... ....................... 13IV.1. MGW CARD TECHNICAL SPECIFICATIONS ........................................... ................................. 13IV.2. FUNCTIONS OF A MGW CARD ....................................... ............................................ .............. 15IV.3. INTRODUCING the MGW card to the system.... ............................................ ............................. 15IV.4. MGW PROGRAMMING THROUGH KNEE............................................... .................................. 17IV.4.1. NETWORK SETTINGS...................................................................... ....................................... 17IV.4.2. CAPABILITY TABLE.................................. .......................................... ..................................... 17IV.4.3. PARAMETERS..................................... ........................................... ......................................... 17IV.4.4. WRITE TO EEPROM................................ ........................................... ..................................... 17IV.4.5. MORE OPTIONS ON KNEE............. ............................................. ........................................... 17

V. IP EXTENSIONS and SIP_SPC MODULE .................................... ........................................ ......... 18V.1. DEFINING IP EXTENSION NUMBERS IN THE SYSTEM ................................................. .......... 18V.2. CONFIGURING IP TELEPHONES..................................... ............................................ .............. 20V.2. 1. ESSENTIAL Parameters........................................... ............................................. ................... 20V.2.2. ADVANCED Parameters ............................................. ............................................. ................. 21V.3. SIP_SPC MODULE SETTINGS .......................................... ............................................. ............ 23V.3.1. NETWORK SETTINGS....................................................................... ....................................... 23V.3.2. CAPABILITY TABLE................................... .......................................... ..................................... 23V.3.3. PROTOCOL DEFINITIONS............................................... ............................................. ........... 23V.3.4. WRITE TO EEPROM................................. ........................................... ..................................... 23V.4. SIP EXTENSION FEATURES ............................................ ............................................. ............. 24V.4.1. IP RELATED SERVICES.............................. ............................................ ................................. 25VI. SIP_TPC and H323 TRUNK MODULES ...................................... ........................................ ......... 30

VI.1. DEFINING IP TRUNK NUMBERS IN THE SYSTEM................................................................... 30VI.2. SIP_TPC / H323 TRUNK MODULE SETTINGS............................................. ............................. 31VI.2.1. NETWORK SETTINGS...................................................................... ....................................... 31VI.2.2. REGISTRATION PROXY .......................................... ............................................. .................. 31VI.2.3. ADDRESS TABLE ..................................... .......................................... ..................................... 31VI.2.4. CODEC TABLE .................................... ........................................... ......................................... 31VI.2.5. WRITE TO EEPROM................................ ........................................... ..................................... 31VI.2.6. MORE OPTIONS ON KNEE............. ............................................. ........................................... 31VII. MAINTENANCE ....................................... ........................................ ......................................... .... 32VII.1. ETHEREAL – WIRESHARK USE........ ............................................. .......................................... 32VII.2. SIP LISTENER USE ......................................... ........................................... ............................... 37VIII. SECURITY ........................................ ........................................ ........................................ ............ 39


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I. INTRODUCTION

More than billions of people are connected to a network for voice or data transmission. This entirenetwork structure includes many kinds of telecommunications and switching interfaces like publicexchanges, routers, hubs, switches, satellites…

This variety of interfaces automatically brought the need to combine all networks under a maininfrastructure, with an acceptable cost. This is where IP communication comes in with its followingoutstanding features:

• Data and voice can be transferred over the same infrastructure.

• IP communications can use the already available infrastructure for communication betweendifferent locations. Since communication does not take place over conventional telephonenetwork, the high cost of long distance communication is virtually eliminated

DS series systems can respond to growing IP need as explained in the next sections.


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II. CONTROLS BEFORE SYSTEM PROGRAMMING

This section gives steps that you need to control before configuring IP modules on Karel system.These steps are independent from Karel systems but they are directly affecting the system’soperation. Cooperation with an administrator who has control on / knowledge of the network isnecessary in these steps.

II.1. CONTROL STEPS FOR BASIC COMMUNICATION

The parameters mentioned in this section are the primary settings to control. If they are notprogrammed properly for telephones, lines and gateways, the communications will not start properly.Sections dedicated to each of these modules contain the related programming details.

IP ADDRESSES OF MODULES

Each module that will be involved in the network must have a unique IP address within the network. If a DHCP server is available on the network, IP telephones can be programmed to get IPaddresses from this server.

MGW and VoIP cards do not have DHCP support on the other hand. So these modules mustbe programmed appropriately as per their static IP addresses.

I I mm p p oo r r t t aa nn t t :: DDSS SS ee r r i i ee ss ss y y ss t t ee mm ss aa r r ee nn oo t t aa bb l l ee t t oo aa c c t t aa ss aa DDH H C C P P ss ee r r v v ee r r .. T T oo uu ss ee t t hh eeDDH H C C P P c c l l i i ee nn t t f f uu nn c c t t i i oo nn oo f f aa mm oo d d uu l l ee c c oo nn nn ee c c t t ee d d t t oo t t hh ee ss y y ss t t ee mm ,, aa ss ee p p aa r r aa t t ee DDH H C C P P ss ee r r v v ee r r i i ss r r ee qq uu i i r r ee d d oo nn t t hh ee nn ee t t w w oo r r k k ..

SUBNET MASK

The IP addresses of the modules connected to Karel system must fall within the same subnet of the

default gateway (e.g., router) of the LAN.GATEWAY ADDRESSThe IP address of the gateway on LAN must be learnt and programmed on the modules connected toKarel system.

GATEKEEPER / PROXY SERVERGatekeepers or SIP Proxy Servers offer extra network management facilities to registered clients.Some of these facilities are:

Numbers dialed by extensions are routed to the IP addresses (the clients do not have todecide routes)

Authentication Bandwidth control

If a gatekeeper exists on the network, EX200 (4VoIP/0), EX200 (8VoIP/0) and EX200 (16VoIP/0)cards with H323 protocol can be programmed to use this gatekeeper. Similarly, the cards with SIPprotocol can first register to a SIP Proxy Server and make the calls over this server.

NAT USEIf NAT (Network Address Translation) is used on the network and a single IP address is used as theInternet access point of network, the modules connected to Karel system must be programmedaccordingly. This access point IP address is referred as NAT IP Address.

STUN UseIf your NAT IP address is not static, you can activate STUN on Karel modules. The advantage

of STUN server is not to care about the dynamic (changing) NAT IP address. STUN server willbe giving the appropriate NAT IP address to the module. Thus NAT IP address does not haveto be re-programmed each time the IP address is changed.


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Important: The efficiency of STUN depends on functionality of STUN Server. If there is afailure on STUN Server, communication will not be established due to change of NAT IPaddress. So if reliable functionality of STUN server cannot be guaranteed, it is recommendedto obtain a static IP addresses for VoIP communication through modems/router etc. and thusto use static NAT IP address settings.

RTP Proxy UseWhen NAT is used within the network, speech and signaling ports must be separatelyforwarded to each IP module’s address to establish communication with those IP modules. If the number of IP modules is high, port forwarding for all IP Modules will be a long processwhich is also subject to mistakes. To get over these difficulties, you can activate RTP Proxyuse on your system. In this case, IP modules will not talk from end-to-end with each other;instead the speech will be carried through RTP proxy and port forwarding will not be needed.Note: RTP Proxy use for IP extensions can be activated on extension basis from IDEA –Virtual Subscriber settings. On the other hand, KNEE address table is used to activate RTPproxy use for IP trunks, on route basis.

DNS USEDepending on the network configuration, Karel modules may be in communication with an external

server to maintain IP communication. If DNS server is activated on the modules, instead of the IPaddress of these external servers, their names can be entered on the modules. In such a case, it willbe the DNS server that will convert these names to IP addresses.

VLANVLANs are logical segments within a corporate LAN. If such segments are available on the networkand there are IP telephones with 2 ports (primary and secondary) for packet communication, theseports can be allocated to different VLANs depending on the packet type: voice signal / data. This willreduce the load on the network and bring efficiency and speed to the communication.

SIGNALLING (TRANSPORT) TYPEIP modules connected to DS series systems can use UDP (User Datagram Protocol), TCP

(Transmission Control Protocol) or TLS (Transport Layer Security) for signaling. If the module is aVoIP gateway, the only signaling type is UDP and it cannot be modified. For IP trunks and extensions,any of these three signaling types can be selected based on the following items:

UDP is commonly used due to its speed: there is no flow control or error correction. This is themain difference between UDP and TCP. TCP protocol offers a guaranteed delivery due to flowcontrol mechanism.

The connected parties may ask you for TCP connection. In that case you can activate TCPsignaling on an IP extension or trunk.

An IP extension that uses UDP signaling cannot call an IP extension with TCP/TLS signaling.Vice versa is also true.

The signaling that will be used in IP trunk applications is flexible and following criteria areused:

- The initial communication is started with the signaling programmed as “SIP TransportType” in KNEE program – “Protocol Definitions” menu.

- If IP trunk is registered to a Proxy Server, the signaling that will be used during registrationcan be programmed in “Registration Proxy” menu. The default value of this parameter is =signaling programmed under “Protocol Definitions”.

- Each route in Address Table can have its own signaling type. The default value of thisparameter is = signaling programmed under “Protocol Definitions”.

- Incoming requests with UDP signaling are always accepted.- If TLS or TCP is selected as signalling type at least once in at least one of the “Protocol

Definitions”, “Registration Proxy” or “Address Table” menus, the calls with TLS / TCPsignaling are also accepted.


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- Another criteria controlled in incoming call requests is the signaling port number. In allsignaling types, the call is not accepted if the signaling port number of that call does notmatch the programmed signaling port number in “Protocol Definitions” menu.

The following examples are given with the assumption that signaling port numbers are inmatch:

Example 1:- Signaling in Protocol Definitions: TCP- Signaling in Proxy Settings: UDP- Signaling in Address Table: UDP.Calls are initiated with UDP signaling. Incoming requests with UDP, TCP and TLS signalingare accepted.

Example 2:- Signaling in Protocol Definitions: UDP- Signaling in Proxy Settings: TCP- Signaling in Address Table: TLS.Registration to the Proxy Server is made with TCP signaling whereas access to the target IP

addresses is made with TLS. Incoming requests with UDP, TCP and TLS signaling areaccepted.

Example 3:- Signaling in Protocol Definitions: UDP- Signaling in Proxy Settings: UDP- Signaling in Address Table: UDP.

All communication is made with UDP signalling; other signaling types are not accepted.

Among these signaling types, TLS is the safest. It provides authentication and encryption of the SIP signaling. The communication is started after a handshake, where various parametersare agreed for the sake of security. The communication with TLS signalling can be monitored

by sniffers but cannot be decoded.

When sRTP is used, TLS is the recommended signaling type. Otherwise, a sniffer can detect SIPlogs of the communication and detect the encryption method & encryption key.


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II.2. CONTROL STEPS FOR VOICE QUALITY

The parameters mentioned in this section are not essential for communication establishment but theyare important to have a high quality communication.

TYPE OF NETWORKThe type of network (being managed or unmanaged) is a key for communication quality.

A managed network (Frame Relay, Virtual Private Network – VPN, Leased Line, etc.) isconvenient for VoIP implementations.

An unmanaged network (Internet-ADSL, Internet-VPN) is not recommended for VoIPcommunications due to negative effects on speech quality caused by delays and packetlosses.

BANDWIDTH

It is important to check the available bandwidth on your network before activating IP modules on your DS200 system. If the network is already loaded with other applications (e-mail servers, webapplications, etc.) and if the required bandwidth amount for IP communications is beyond network’slimit, both IP communications and those other applications will suffer decrease in performance,especially in heavy traffic.

A table is given in the next pages to make a bandwidth evaluation more easily.

TOSTOS parameter in Karel IP lines and gateways can be programmed to desired values. If the router of your network is capable of giving a priority based on these TOS values, speech quality can beimproved since the router will give higher priority to voice packets and lower the rate of losses anddelays.

Note: This facility depends on specifications of network router.

ECHO CANCELLATIONThe echo in IP communications can be a real trouble due to the topology of the IP networks. Thedelays in the communication, the impedance mismatch of analog ports may cause echo on IP calls.DS200 systems are equipped with advanced echo canceler algorithms and thus echo is not a problemfor DS200. But yet huge delays may decrease the performance of echo canceler so the users may stillhear echo. In such cases, VLAN and other ToS solutions must be considered by the networkadministrators.

PACKET CONTROL FEATURESKarel IP telephones, lines and gateways have packet controlling features like dynamic jitter buffer (controls delay variations that occur due to different paths traced during transmission) , PLC algorithm(controls packet loss). If other devices on the network support these features as well, they will make apositive effect on quality of communications.

Voice Activity Detection (VAD): Being an important packet control feature, it helps to detect silentperiods during a call. Those detected silence periods are not sent to the network, thus bandwidth isprotected. This feature can be supported by the CODEC G729A and must be used at both sides of theconnection.


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II.3. CONTROL STEPS FOR NETWORK DEVICES

The parameters mentioned in this section are the controls on the modules that lie within the samenetwork as Karel systems.

PERMISSIONSIf communication between different networks is also required and conversations with parties atdifferent ends will be carried out; firewalls, routers and similar network access points have to beprogrammed appropriately at each network by network administrators. This programming is nothingbut port forwarding on these devices to the relevant IP modules. A table is given below for thispurpose:

Application Name / Explanation TCP / UDP Port Number to beforwarded

Destination IPaddress

For signaling between SIPextensions/lines/gateways and WAN,

incoming requests to this port must beforwarded to local IP address of thetelephone/line/gateway /MGW

UDP /TCP/TLS 5060 /5060/5061 IP address of telephone, line,

gateway or MGW

For signaling between H323lines/gateways and WAN, incomingrequests to this port must be forwarded tolocal IP address of the gateway

TCP 1720 Line or gateway’s IPaddress

For conversations between SIP telephonesand WAN, incoming requests to this portmust be forwarded to local IP address of the telephone

UDP Changes wrttelephone model

Telephone’s IPaddress

For conversations between SIP telephones

and WAN, incoming requests to this portmust be forwarded to local IP address of MGW

UDP 54000 MGW IP address

For conversations between SIP / H323gateways and WAN, incoming requests tothis port must be forwarded to local IPaddress of gateway

UDP 5004 Gateway IP address

For KNEE connection to Karel modules UDP 14200 Karel module’s IPaddress

For KNEE connection to Karel modules TCP & UDP 14201 Karel module’s IPaddress

For SIP Listener connection (maintenance) UDP 12120 Karel module’s IPaddress

SWITCH TYPESTo have a qualitative communication, layer 2 or higher switches are recommended. (Hubs increasethe load on the network and this may lead to decrease in speech quality.)

Note: MGW2 cards do not work with GB switch and 10 Mbit hub.

CABLE TYPESDue to their capacity and isolation quality, CAT 5 or higher cables must be preferred.

Note: Unexpected problems like packet loss can be faced with handmade network cables.


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II.4. BANDWIDTH USAGE BY KAREL MODULES

Currently G.711 and G.729 are the supported CODECs in Karel systems and the required bandwidthdepends which one of these is used. If MGW2 card is used, G.723 and iLBC are supported as well

The key points that will help to decide which one of these CODECs will be used are:

• G.711 uses 64 kb to transfer 1 second of data whereas G.729 uses 8 kb to transfer the sameamount. This value is 5.3 kb or 6.3 kb for G.723 and 15.2 kb (for frames of 20 sec) or 13.33 (for frames of 30 sec) kb for iLBC.

• Being independent from the CODEC used, each packet contains a header of 54 bytesapproximately.

If as an example 100 voice packets are transferred in a second, just 43.2 kbits (100 voice packetsrequire 5400 bytes in total and considering 1 byte = 8 bits, this will be 43.2 kbits) will be used byheaders. This will require 51.2 kbps with G.729 whereas 107.2 kbps will be required by G711.

The following table shows the required bandwidth compared to G.729 and G.711 with respect todifferent packet sending intervals.

Packetsendingintervals

Codec Number of packets in asecond

Voice in a second Headers in asecond

Total

G.711 100 64 kb 43.2 kb 107.2 kbpsG.729 100 8 kb 43.2 kb 51.2 kbpsG.723 100 5.3 kb or 6.3 kb 43.2 kb 48.5 kbps or

49.5 kbps

10 ms

iLBC 100 15.2 kb or 13.33kb

43.2 kb 58,4 kbps or 56,53 kbps

G.711 50 64 kb 21.6 kb 85.6 kbpsG.729 50 8 kb 21.6 kb 29.6 kbpsG.723 50 5.3 kb or 6.3 kb 21.6 kb 26.9 kbps or

27.9 kbps

20 ms

iLBC 50 15.2 kb or 13.33kb

21.6 kb 36,8 kbps or 34,93

G.711 25 64 kb 10.8 kb 74.8 kbpsG.729 25 8 kb 10.8 kb 18.8 kbpsG.723 25 5.3 kb or 6.3 kb 10,8 kb 16,1 kbps or

17,1 kbps

40 ms

iLBC 25 15.2 kb or 13.33kb

10,8 kb 26 kbps or 24.13 kbps

Required Bandwidth for a DS200 system= No. of IP extensions × Required Bandwidth for the used CODEC by telephones +No. of channels of MGW card × Required Bandwidth for the used CODEC by MGW card + no of IPtrunk channels x required bandwidth for used codes on trunks.

ExampleNo. of IP extensions: 20No. of IP trunks :10No. of channels of MGW card: 16

Codec: G.729Packet sending interval: 40 msecRequired bandwidth = [20 x 18.8 kb] + [16 x 18.8 kb] + [10 x 18.8 kb] = 864.8 kbps


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II.4. THE LIST OF KAREL MODULES

In the following pages the following Karel IP modules will be explained:

1) VOIP (Voice Over IP) Gateway Module

2) MGW (Media Gateway - IPTDM converter) Module

3) IP extensions and SIP_SPC (IP Extension software) Module

4) IP trunks and SIP_TPC (IP Trunk Software) and H323 trunk modules

The programming of these modules requires different media: IDEA, telephone’s own Web interfaceand KNEE. The steps are given with basic information assuming the fact that the programmer isalready familiar to the mentioned programming tools.


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III. VoIP GATEWAY

The following sections explain VoIP Gateway card with its different aspects:

- Its technical specifications

- Its functions

- Its programming in two phases: programming on the system, programming on VoIP Gatewaycard.

VoIP Gateway card is used as an IP trunk card on DS200 systems which does not have IP enabledCPU cards.

III.1. VOIP GATEWAY CARD TECHNICAL SPECIFICATIONS

• There are three different VoIP Gateway cards: EX200 (0/4VoIP), EX200 (0/8VoIP) and EX200(0/16VoIP). The only difference between cards is the number of channels. Their programmingdetails are identical.

• Each VoIP Gateway card can have either H323 or SIP protocols.

• A VoIP gateway module actually consists of two cards: a main board and a CPU card that lies ontop of this main board. The CPU card is identical for all VoIP gateway cards. The main board onthe other side is different as per the number of channel resources.

• There are two software files on a VoIP gateway module: main software and u-boot software.

- U-boot software is unique for all VoIP gateway modules.

- There are 6 different main software files: the software files for EX200 (0/4VoIP), EX200(0/8VoIP) and EX200 (0/16VoIP). Additionally, the software files that supportH323 and SIPprotocols are different for each capacity.

• The main software file can be programmed through KNEE program or through a direct PCconnection via serial port of the card. KNEE offers the option to load u-boot software but it isrecommended to load this software through direct PC connection via serial port of the card for amore reliable connection.


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III.2. FUNCTIONS OF A VOIP GATEWAY CARD

Main function of VoIP Gateway card is to establish connection between your DS200 system and the IPnetwork. While performing this basic function, the card can be used for two basic functions:• As a direct gateway (without a proxy server) between your exchange and IP network. This use can

be thought of as an IP tie-line connection.

• First the card registers to a Proxy Server and the calls to IP network all forwarded to that proxy.

III.3. PROGRAMMING ON THE SYSTEM

The system-side programming for a VoIP gateway card is basically the number routing, namely LCR.The details of LCR programming are given in DS Series Programming Guide. In this guide, just afigure is given below:

The remote system extension numbers (in the range 1525 and 1566) are routed to the line 7200,without any LCR filtering. 7200 is the port number of VoIP gateway card on slot 58.


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III.4. PROGRAMMING ON VOIP GATEWAY CARD

IP related settings of VoIP Gateway cards are made by KNEE software. KNEE software includesdifferent parameters in its menus. The following sections are separated as per the KNEE menus.

This section explains the essential parameters for operation of VoIP gateway cards. For other adjustments, which are rather details of applications, KNEE guide must be referred.

KNEE programming options can vary depending on protocol of card being SIP or H323. Theprogramming options are explained with the note of the supported protocol.

III.4.1. NETWORK SETTINGS

(Valid for SIP – H323 protocols)

This is the place where primary network settings of VoIP gateway card are made.a) The following settings are essential for the proper operation of VoIP Gateway cards:

- IP address

- Gateway address

- Subnet mask

b) If WAN access is required in the communication, NAT IP address must be programmed.

c) The following settings may be programmed if required:

- TFTP server

- DNS & STUN settings (for cards with SIP protocol)

- Gatekeeper (for cards with H323 protocol)

III.4.2. ADDRESS TABLE

(Valid for SIP – H323 protocols)

Address table is the place where numbers dialed by extensions are checked and thus the correct IPaddress to forward the call is resolved.

The following fields of this table must be filled out to have a proper IP routing:

- Numbers

- Destination IP address

III.4.3. CAPABILITY TABLE

(Valid for SIP – H323 protocols)This table is used to program the codec types that will be used in IP communications and is very critical for the sake of communication.

I I f f t t hh

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I I P P

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d d

d d

r r ee

ss

ss

d d

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nn

oo

t t m

maa

t t c c

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i i t t hh

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OO

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ss

oo

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V V

oo

I I P P

g g aa t t ee w w aa y y c c aa r r d d ,, v v oo i i c c ee w w i i l l l l nn oo t t bb ee t t r r aa nn ss mm i i t t t t ee d d bb ee t t w w ee ee nn p p aa r r t t i i ee ss ..Note: Always use the Capability as “Rcv&XmtAudio”.


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III.4.4. PARAMETERS

(Valid for SIP – H323 protocols)These parameters control communication specific parameters like receive – transmit voice levels, packetsending intervals, TCP connection settings and delays in passing the packets coming from network.

The following parameters in this menu are important:

- Gain levels

- Fast start. (Available only with H323 protocol.) It must match to the settings of destination.Otherwise, call request will not be taken into account.

- NAT IP address. It must be programmed for cards with H323 protocol if WAN access is required.(Cards with SIP protocol require this setting in Network Settings Menu.)

- RTP_UDP port: 5004 by default. Programming it may be necessary if WAN access is problematicat speech side.

- Signaling port: 1720 for H323 and 5060 for SIP by default. Programming it may be necessary if WAN access is problematic at signaling side.

III.4.5. REGISTRATION PROXY

(Valid for SIP protocol)This menu includes parameters required for registration to a Proxy Server. If registration to a server is notrequired, you do not have to use this menu at all. One card can be registered to multiple servers at a time.

For mote details on registration to a proxy server, please refer KNEE manual. Please also check thesettings in the “Endpoint Settings” menu.

III.4.6. WRITE TO EEPROM

This option saves the recent programming changes on the memory of VoIP Gateway card. If not used, thelast settings will not be recalled after the card is powered OFF and ON.

III.4.7. MORE OPTIONS ON KNEEKNEE program offers more programming options for VoIP Gateway cards. The details of these menus canbe found in KNEE manual. Below, their brief descriptions are given.

a) Protocol definitions (valid for SIP protocol): Detailed parameters like T1- T2 retransmission times,session expiring durations, DTMF settings can be programmed.

b) Routing settings (valid for SIP protocol): Alternative routes for various erroneous cases (like Ping or QoS errors) can be programmed.

c) KNE user accounts (valid for SIP protocol): Different user accounts can be defined to program or view VoIP Gateway cards.

d) Manipulation tables (valid for SIP protocol): Called and calling party numbers can be modified if thedesired extension makes a call and/or a desired number is called. The modification is to add or delete digits from the front or end.

e) Status and diagnostic (valid for SIP protocol): KNEE also offers basic monitoring options.

f) Software update: Recommended only for upgrade of main software. This option should not beused for u-boot software upgrade.


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IV. MGW CARD PROGRAMMING

The following sections explain MGW card with its different aspects:

- Its technical specifications

- Its functions

- Its programming in two phases: programming on the system, network programming on MGW card.

IV.1. MGW CARD TECHNICAL SPECIFICATIONS

There are two types of MGW cards: MGW and MGW2.

MGW card:

MGW and EX200 (16 VoIP/0) cards are identical except for their main software and their appearance onIDEA Configuration window. The following items are comparison of these two cards:

• Their cards are identical in hardware

• Their CPU cards are identical (the small cards that lie on the main cards)

• Their u-boot software is identical

• Their software upgrade methods are identical (either by IP connection over KTFTP server or HyperTerminal connection through serial part)

But,

• Their main software is different.

• Their appearances are different.

MGW2 card:

MGW2 and MGW cards have different hardware/software and they are different in terms of technicalspecifications as well.

• MGW2 card has two slots for blackfin cards. Each blackfin card can have one or two DSP chips. EachDSP chip has 12 speech channels; hence a single MGW2 card can support 48 speech channels. Thisnumber is 16 for MGW card.

• MGW2 communicates with master software over Ethernet.• MGW card supports G.711, G.729 and G729AB CODECs whereas MGW2 card supports G.711,

G.729, G729AB, G723, iLBC and T38.

• MGW card does not support encrypted speech, namely sRTP whereas MGW2 card supports thesame. Usage of sRTP is subject to authorization and 2 speech channels are used in sRTP speeches.(48 channel of MGW2 card can be used when sRTP is used.)

• DSP chips on MGW2 card can be configured as RTP proxy. Thus signaling and speech will be madethrough different servers.


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IV.2. FUNCTIONS OF A MGW CARD

Main function of MGW cards is to carry speech between IP and non-IP components of Karel system.

For MGW card, communication with non-IP components is established over the PCM channels of backplane. For MGW2, all communication is made over Ethernet.

(IP components: IP extensions, IP lines

Non-IP component: All of the remaining components: analog/digital extensions/lines, IP gateways.)

Additionally, location of MGW card on the system is used as the node of IP extensions / lines. Therefore,before introducing IP extensions and IP lines to the system, it is necessary to introduce a MGW card tohave speech between IP and TDM ports.

IV.3. INTRODUCING THE MGW CARD TO THE SYSTEM

Once the MGW card is installed on the system, it is necessary to specify its slot location in following steps.

• Go to “Virt. Sbsc. Settings” menu on IDEA.

• Press “Branch Exchange Node Table”.

• Enter the starting – ending port numbers of MGW card and select the application that will use thisMGW card as a node: “SIP” for IP extensions, “IP trunk” for IP lines. (If the same card will be used both

for SIP extensions and trunks, the same port numbers must be defined at different rows as SIP and IPtrunk.)


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H H I I N N T T :: SS t t aa r r t t i i nn g g aa nn d d ee nn d d i i nn g g p p oo r r t t nn uu mm bb ee r r ss oo f f aa M M GG W W c c aa r r d d c c aa nn bb ee ss ee ee nn bb y y

p p oo i i nn t t i i nn g g t t hh ee mm oo uu ss ee p p oo i i nn t t ee r r oo nn i i t t i i nn C C oo nn f f i i g g uu r r aa t t i i oo nn w w i i nn d d oo w w ..


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IV.4. MGW PROGRAMMING THROUGH KNEE

IP settings of a MGW card are made over KNEE program, just like a regular VoIP Gateway card.

IV.4.1. NETWORK SETTINGSProgram the following fields on this menu:

- IP address

- Gateway address

- Subnet maskIf you want to upgrade MGW card software through KNEE, also program TFTP server IP address.

IV.4.2. CAPABILITY TABLE

This table is used to program the codec types that will be used in IP communications and is very critical for the sake of communication between IP and non-IP modules of the system:I I f f t t hh ee I I P P c c oo mm p p oo nn ee nn t t ss t t hh aa t t w w i i l l l l bb ee i i nn c c oo mm mm uu nn i i c c aa t t i i oo nn w w i i t t hh t t hh ee ss y y ss t t ee mm hh aa v v ee d d i i f f f f ee r r ee nn t t C C OO DDE E C C ss f f r r oo mm t t hh ee C C OO DDE E C C ss oo f f M M GG W W c c aa r r d d ,, t t hh ee c c oo mm mm uu nn i i c c aa t t i i oo nn bb ee t t w w ee ee nn N N oo nn --I I P P aa nn d d I I P P

p p oo r r t t ss w w i i l l l l nn oo t t bb ee ee ss t t aa bb l l i i ss hh ee d d .Note: On MGW card, always use the Capability as “Rcv&XmtAudio”.

IV.4.3. PARAMETERSThese parameters control communication specific parameters.

The following parameters in this menu are important:

- Gain levels

- RTP_UDP port: 5004 by default. Programming it may be necessary if WAN access is problematicat speech side.

IV.4.4. WRITE TO EEPROM

This option saves the recent programming changes on the memory of MGW card. If not used, the lastsettings will not be recalled after the card is powered OFF and ON.

IV.4.5. MORE OPTIONS ON KNEEThe details of other KNEE menus can be found in KNEE manual.


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V. IP EXTENSIONS and SIP_SPC MODULE

The following sections explain the programming steps of IP extensions:

- Introducing IP extension numbers to the system

- Configuring IP telephones to connect to the system

- SIP_SPC module settings.

P P r r i i oo r r t t oo t t hh ee ss ee p p r r oo g g r r aa mm mm i i nn g g ss t t ee p p ss ,, aa M M GG W W c c aa r r d d t t hh aa t t i i ss p p r r oo g g r r aa mm mm ee d d aa ss SS I I P P nn oo d d ee t t y y p p eemm uu ss t t bb ee aa v v aa i i l l aa bb l l ee oo nn t t hh ee ss y y ss t t ee mm – – r r ee f f ee r r t t hh ee p p r r ee v v i i oo uu ss ss ee c c t t i i oo nn ..

V.1. DEFINING IP EXTENSION NUMBERS IN THE SYSTEM

IP extension numbers can be defined from “Virt. Sbsc. Settings” menu on IDEA. It is necessary to fill outfollowing fields:

Access code: The access code of the SIP extension (the telephone number of the IP phone).

Node number: The index for the MGW card. This is the row number on “Branch Exchange Node Table”where MGW location is programmed as SIP.

IP Number Usage: If this is programmed as “changeable”, any IP telephone can register to the system withthis access code. If it is programmed as “fixed”, the system requires the telephone to have the IP addressas programmed in “IP #” field.

SIP password: IP extensions are required to have a password during registration. The default value of thispassword is “abcdefgh12345678”.

Authentication Control: If programmed as “at registration”, password from an IP extension is required onlyduring its registration. If programmed as “at registration and incoming calls”, password is required in eachcall attempt as well.

RTP Proxy Mode: RTP Proxy use for an IP extension is programmed within this field. When it is adjustedas “automatic”, RTP Proxy will be used only if the IP extensions is registered behind NAT (if it is an internalIP extension RTP Proxy is not used). Other programmable options are “always” and “never”.

Other fields in this table are for maintenance purposes. They cannot be edited.

Port #: The SIP signaling port number of the extension. This is entered from telephone and the systemprogramming does not have any effect.

Registered :: Shows if the IP extension with the given access code is registered to the system or not.

IP#: IP address of the phone. Must be entered if the “IP Number Usage” parameter is chosen as “Fixed”.The IP address of the IP phone is automatically obtained during registration and written here if the sameparameter is chosen as “Changeable”.


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R R E E GG I I SS T T R R A AT T I I OO N N P P R R I I N N C C I I P P LLE E SSIt might be useful to list down some registration principles of the system to visualize the system capabilitiesbetter:

• The system does not register multiple IP addresses with the same access code.

• In case of a register request from an IP extension the system checks following points:

- Is the telephone number defined as an extension number? If no, the request is denied. If yes,

- Does the request contain correct password? If no, request is denied. If yes,- (If IP number usage: fixed) Does the request come from correct IP address? If notrequest is denied.

If the answers to all of these points are YES, then the IP extension is registered to the system.

• The IP extensions can stay registered to the system for a certain period. This period is calledREGISTRATION PERIOD and it is programmed from the telephone. At the end of this period,registration is terminated by the DS200 system.

If registration period is too short, registration process will cause extra traffic on the system. I theduration is too long, the system may try to reach this extension considering that it is still registeredeven when the Ethernet connection does not exist. So, an optimum value must be used here.

• Within registration period, the system keeps access code, node number, fixed/changeable IP use, SIPsignaling port number, password, authentication control method and register status of the IP extensionin its memory. If the system is required to re-start within this registration period, the extension will be re-registered from these settings.

• Once an IP telephone is registered to the system, it becomes an extension and its parameters can beprogrammed through IDEA just like a TDM extension.


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V.2. CONFIGURING IP TELEPHONES

V.2. 1. ESSENTIAL PARAMETERS

In order to register to a Karel system, the following adjustments are necessary on an IP telephone:

SIP server address: The IP address of the Karel system must be programmed as SIP server of thetelephone. This IP address is the IP address of the Ethernet port on CPU card of your system which isused by the SIP_SPC software.

SIP ID: The access code that will be used as extension number must be programmed as SIP ID of thetelephone.

SIP (authentication) password: The password defined in “Virt. Sbsc. Settings” menu (which isabcdefgh12345678 by default) must be programmed as SIP password.

CODECs: At least one CODEC used by the telephone must match with a CODEC from MGW card’s list.

Example from NT32I:1. Write the IP address of the telephone on Web browser and open Web interface of the telephone.


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2. Go the protocol settings and enter the required parameters.

3. Set CODECs.

V.2.2. ADVANCED PARAMETERS

Depending on the requirements some extra settings might be necessary as well. These settings will appear during use of the telephone as an extension of the system. Some might be required due to network settingswhereas some might be required by the system. The most common ones are listed below:

SIP and RTP port numbers: These port numbers are used if a call comes from WAN to this IP telephone.The firewall – router (or similar network device) must be programmed such that the call requests for thisport must be forwarded to the IP address of this telephone.


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Register period: This is the duration that the system will store parameters of the IP telephone in itsmemory.

DTMF settings: If you want to use Leave Message feature of your system, you must activate “In-bandDTMF” setting. This might appear as “SIP info” in some telephones.

PRACK: This is a standard SIP feature that stands for acknowledge for provisional messages (like ringing).

It does not require any extra setting on Karel system side.User Name: This name is shown in the calls between two IP telephones. If another name is given throughIDEA – port parameters, IDEA name will be used.

I I M M P P OO R R T T A A N N T T ::F F oo r r ss oo mm ee SS I I P P t t ee l l ee p p hh oo nn ee ss ,, p p r r ee ss ss i i nn g g “ “ SS aa v v ee ” ” oo r r “ “ OO K K ” ” i i ss nn oo t t ee nn oo uu g g hh t t oo aa c c t t i i v v aa t t ee

p p r r oo g g r r aa mm mm i i nn g g c c hh aa nn g g ee ss .. T T hh ee y y mm aa y y aa l l ss oo r r ee qq uu i i r r ee t t ee l l ee p p hh oo nn ee r r ee ss t t aa r r t t .. E E x x i i ss t t ee nn c c ee oo f f ss uu c c hh aar r ee qq uu i i r r ee mm ee nn t t mm uu ss t t bb ee c c hh ee c c k k ee d d f f r r oo mm t t hh ee t t ee l l ee p p hh oo nn ee ’ ’ ss mm aa nn uu aa l l ..


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V.3. SIP_SPC MODULE SETTINGS

Depending on your network requirements or specifications, you may need to activate extra settings on your system. These IP related settings are saved in configuration file, which is named as SIP_SPC.CONF.

Again, KNEE is the interface to enter settings to the SIP_SPC.CONF file.

V.3.1. NETWORK SETTINGSThe IP address of the CPU card must be entered. Normally this will be same IP address with the one youhave entered while running KNEE program.

If WAN access is required in the communication, NAT IP address must be programmed.

DNS & STUN settings can be programmed if required.

V.3.2. CAPABILITY TABLE

This is nothing but a list that contains list of CODECs used in IP communications in your system.

V.3.3. PROTOCOL DEFINITIONSDetailed parameters like T1- T2 retransmission times, PRACK usage, session expiring durations, DTMFsettings can be programmed.

V.3.4. WRITE TO EEPROM

This option is required to save the recent programming changes. If not used, the last settings will not berecalled after the system is powered OFF and ON.


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V.4. SIP EXTENSION FEATURES

SIP extensions can use various features as extensions of your system. Some of these features are theones defined by SIP standards, whereas some of them are facilities of Karel DS series systems. Inother words, some of the usable features can only be used when SIP telephones are registered toKarel DS series systems.

Some of DS series features, which can also be used by SIP extensions, are listed below:

- Marked External Call - Call Back

- Account Coded Call - Leaving message to other extensions

- Forced Account Coded Call - Parked Call Retrieve

- Marked and Forced Account Coded Call - Conference

- Changing forced account code password - Listening to messages

- Password dialing from another extension - Door opener

- Chief-Secretary Mode - Name display

- Last Number Auto-Dial - PBX group log-in / log-out

- Redial - Call pick up

- Password define/change - Group call pick up

- Phone lock - Group external call pick up

- External calls from locked phone - Line-line connection

- Reminder Service (with – without message) - Time/date setting

- Temporary – Permanent Absent Message - Alarm service

I I M M P P OO R R T T A AN N T T ::

SS I I P P t t ee l l ee p p hh oo nn ee ss ss uu p p p p oo r r t t f f ee aa t t uu r r ee ss oo nn l l y y i i nn N N --bb l l oo c c k k d d i i aa l l i i nn g g mm oo d d ee ..

Examples:

1. In order to define the password as 1234, SIP telephone must first enter the digits 8361234 andthen send these digits.

2. To start an external call using the account code 116, SIP extension must enter the digits 797116and then send these digits.

For some of the DS series features, activation methods of SIP extensions are slightly different from themethods of TDM extensions. These facilities are noted below:

LEAVING MESSAGE TO OTHER EXTENSIONS

SIP telephones cannot leave message to a ringing extension. They can leave messages to other extensions only by pressing 6 after receiving Busy or No Answering menu of ACD. Both the calledextensions and calling SIP extensions must have the necessary permissions to use EVM menus.

CONFERENCE

SIP telephones can start conference by first pressing * and then putting the call on hold, as shown inthe sequence below


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- Call first party.

- Press *. The speech will continue.

- Hold the call with the respective key on the phone.

- Call second party.

- After talking to second party, press *. Conference will start. To include more extensions you canhold the ongoing conference, and include more parties with *.

CALL BACK

SIP telephones can use this feature as follows:

- To activate Call Back: 811 + access code

- To deactivate Call Back: 810 + access code

When Call Back call will be realized, while the called party is ringing, SIP extension is kept on hold. Assoon as called party answers the call, connection with SIP extension is realized.

V.4.1. IP RELATED SERVICES

In addition to the features that can be used by TDM extensions, there are some facilities defined bySIP standards and they can only be used by SIP extensions of your system. These features are givenbelow with related notes to use them.

I I M M P P OO R R T T A AN N T T ::

P P R R E E SS E E N N C C E E aa nn d d I I N N SS T T A A N N T T M M E E SS SS A A GG I I N N GG f f aa c c i i l l i i t t i i ee ss c c aa nn O O N N LL Y Y bb ee uu ss ee d d i i f f t t hh ee ss y y ss t t ee mmhh aa ss t t hh ee r r ee qq uu i i r r ee d d l l i i c c ee nn ss ee .. P P l l ee aa ss ee c c oo nn t t aa c c t t K K aa r r ee l l t t oo hh aa v v ee t t hh ee r r ee qq uu i i r r ee d d l l i i c c ee nn ss ee ss ..

PRESENCEProvided that the system has the required license, IP extensions can see the status of other extensions, if they are programmed accordingly. The system can broadcast 60 presence informationat a time.

Note: For the IP telephone models that do not support presence facility, if BLF lamps are available onthe phone, those LEDs can show status of other ports as Busy / Idle.

- IP extensions send their current status to the system with PUBLISH messages. (Defined by RFC3903.)

- IP extensions ask for the status of other IP messages with SUBSCRIBE messages. The systemthen send the status of extensions to the requesting parties by NOTIFY messages. (Defined byRFC 3265.)

- Once the necessary programming steps are traced, the following status of extension canautomatically be viewed by the requesters:Idle,On the phone,Ringing,Busy (defined by user),

Away (defined by user),Idle (appears automatically after some period defined by the user) andOff-line.

- The IP extensions can also define a message that will appear along with this status. (Defined by

RFC 3863.)

The following figures are given to illustrate Presence activation steps, on a softphone.


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a. First step: Send current status to the system by registering to the Present Agent, which is a part of SIP_SPC.

b. Second step: Ask for status information of extensions. You can ask status of non-IP extensions aswell as IP extensions.

c. Application: See the status of extensions.


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INSTANT MESSAGING

Provided that the system has required licenses, two IP telephones can send messages to each other as per standards defined by RFC 3428.

- The messages that cannot be sent are stored in a database server.

- The default size of this database is 500 Kbytes.

- The size of this database is controlled within each 50 minutes. If the size of the database hasreached 80% of 500 Kbytes, then the older messages are erased until the size is reduced down to50% of 500 Kbytes.

- As the unsent messages are stored in the database, if the target IP extension is not registered atthat moment, he will be able to receive his messages after his registration.

- SIP_SPC.CONF program can be used to control the database, with the following commands. If these commands are not typed in SIP_SPC.CONF file, the default values will be used.

• db_ipaddr: Normally the database server and SIP_SPC are on the same location. If required,

a separate address can be used as database server. In that case, this command must beused to define this address.

• mess_db_size: Size of the database (in KB) is defined with this command. (Default = 500Kbytes)

• mess_db_upper_limit: The highest limit (in %) is defined with this command. (Default = 80%)

• mess_db_lower_limit: The minimum limit (in %) is defined with this command.( Default =50%)

A view of SIP_SPC.CONF file is given below:

VIDEO CALLS

Video calls can be established if the used IP telephones have video support. Any of the video codes -H263, H263-1998, H.264 (low quality),H.264 (high quality) - can be used for video calls.


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SESSION TIMER

Session timer is a method to check the continuity of the calls. If the telephones hang up unexpectedly(disconnection of Ethernet cable, power failure etc.), this can be detected within a session intervalduration and the call can be terminated

When session timer is activated, the participants first agree on the session interval, and then the partythat will control the process is decided. The responsible side sends UPDATE and INVITE messageswithin session interval. If no answer is detected, the call is terminated.

- Session timer can be defined on IP telephones if desired. However, SIP_SPC already uses 300seconds as session interval.

- When both parties have session intervals with different duration, the valid session interval will bethe interval of called party.

- The first UPDATE and INVITE messages are sent at the half of session interval. (e.g. if sessioninterval is decided as 500 msec, UPDATE and INVITE messages are sent at each 250 msec.)

- SIP_SPC.CONF program uses the following commands to control session timer facility. If thesecommands are not typed in SIP_SPC.CONF file, the default values will be used.

• session_Interval: Session interval duration (in seconds) that will be used by SIP_SPC.(Default = 300 sec)

• mins_SE: The minimum value that can be accepted as a valid session interval by SIP_SPC.(Default = 90 sec)

A view of SIP_SPC.CONF file is given below:


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VI. SIP_TPC and H323 TRUNK MODULES

IP trunk application is a better version of VoIP Gateway application. The differences between thesetwo applications can be listed as follows:

• A VoIP gateway application requires a dedicated card (VoIP Gateway card). IP trunk applicationdoes not need any hardware. It uses of MGW card, only when TDM extensions will use IP trunks.

• VoIP gateway cards carry both media and signaling. On the other hand, in an IP trunk application,signaling is made through IP trunk and media is transmitted through MGW (in TDM calls) or directly between IP nodes.

The following sections explain the programming steps of IP trunks.

- Introducing IP trunks to the system

- SIP_TPC module or H323 trunk module settings.

P P r r i i oo r r t t oo t t hh ee ss ee p p r r oo g g r r aa mm mm i i nn g g ss t t ee p p ss ,, aa M M GG W W c c aa r r d d t t hh aa t t i i ss p p r r oo g g r r aa mm mm ee d d aa ss I I P P t t r r uu nn k k nn oo d d eet t y y p p ee mm uu ss t t bb ee aa v v aa i i l l aa bb l l ee oo nn t t hh ee ss y y ss t t ee mm – – r r ee f f ee r r t t hh ee p p r r ee v v i i oo uu ss ss ee c c t t i i oo nn ..

VI.1. DEFINING IP TRUNK NUMBERS IN THE SYSTEM

Defining IP trunk numbers is nothing but to define new access codes in “IP Trunks” section of portslist. The only difference from regular analog / digital lines is just one parameter:

Node number: The index for the MGW card. This is the row number on “Branch Exchange Node

Table” where MGW location is programmed as IP Trunk.


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VI.2. SIP_TPC / H323 TRUNK MODULE SETTINGS

KNEE is the interface to enter extra settings for IP trunks, including network adjustments, addressrouting etc. These settings are almost identical to VoIP gateway cards.

KNEE settings for H323 trunks are made through 1 st ethernet port of CPU card.

KNEE settings for SIP_TPC are made through 2 nd ethernet port of CPU card.

VI.2.1. NETWORK SETTINGSThe IP address of the CPU card must be entered as “Master IP address”. This is the address of thesecond Ethernet port on your CPU card.

If WAN access is required in the communication, NAT IP address must be programmed.

DNS & STUN settings can be programmed if required.

VI.2.2. REGISTRATION PROXYThis menu includes parameters required for registration to a Proxy Server. An IP trunk can beregistered to multiple servers at a time. If registration to a server is not required, you do not have touse this menu at all. For mote details on registration to a proxy server, please refer KNEE manual.Please also check the settings in Endpoint Settings menu.

VI.2.3. ADDRESS TABLE

The routing of dialed numbers to the desired destination IP addresses must be programmed, just likea VoIP Gateway card.

VI.2.4. CODEC TABLEThis is nothing but a list that contains list of CODECs used in IP communications in your system.

VI.2.5. WRITE TO EEPROM

This option saves the recent programming changes. If not used, the last settings will not be recalledafter the system is powered OFF and ON.

VI.2.6. MORE OPTIONS ON KNEEKNEE program offers more programming options for IP trunks. The details of these menus can befound in KNEE manual. Below, their brief descriptions are given.

a) Protocol definitions (valid for SIP protocol): Detailed parameters like T1- T2 retransmissiontimes, session expiring durations, DTMF settings can be programmed.

b) Routing settings (valid for SIP protocol): Alternative routes for various erroneous cases (likePing or QoS errors) can be programmed.

c) KNE user accounts (valid for SIP protocol): Different user accounts can be defined to programor view VoIP Gateway cards.

d) Manipulation tables (valid for SIP protocol): Called and calling party numbers can be modifiedif the desired extension makes a call and/or a desired number is called. The modification is toadd or delete digits from the front or end.

e) Status and diagnostic (valid for SIP protocol): KNEE also offers basic monitoring options.


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VII. MAINTENANCE

The IP system, unlike the conventional circuit switched communication systems, uses commonresources with the data network of the companies. Therefore, the maintenance of a data network andthe voice network have a lot in common.The IP traffic density on the system is related to the bandwidth of the IP infrastructure (as explained inthe previous sections) and to the processing power of the server used for SIP_SPC and SIP_TPCapplications.

On the CPU card of a DS series system with the PPC 850 processor, simultaneously 128x128 IPextensions (256 IP extensions in total) and 100 IP trunks can be used. However, if an external PC isused as the SIP_SPC and SIP_TPC servers, the simultaneous communication channel number increases depending on the processing power of the used server. It is highly recommended tocalculate the required traffic density and depending on this calculation, select an appropriate server for the IP applications.

As the server is setup and run, the run time of the IP system may require some on-line monitoring for

bandwidth utilization, packet loss rate, delays and etc. IT departments generally have such monitoringtools. If not, such tools should be obtained for proper maintenance capabilities.

In case of a problem on any IP module, IP monitoring will be required. Based on these logs Karel willbe able to give assistance. Karel systems offer two tools for monitoring:

• Use of an open source program (Ethereal or Wireshark). This requires connection to themonitored destination over a HUB or programmable switch (with port mirroring feature).

• Use of Karel SIP listener program. This can be used for IP applications with SIP protocol, if a Hubor programmable switch is not available.

VII.1. ETHEREAL – WIRESHARK USE

Ethereal (or Wireshark) is actually a network analyzer. To be able to receive IP packets, the Etherealmust be running on a PC that is in connection to the monitored destination (e.g. Karel MGW card) viaa Hub. If Hub is not available, a programmable switch can also be used. In that case, the PC must beconnected to a port that is listening to other ports on switch. This is called “port mirroring” and asample snapshot for port mirroring is given below.


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If an Ethereal running PC is connected to port 1 of this switch, incidents from ports 3, 4, 5,…8 can bemonitored. An Ethereal log file can be retrieved in following steps:

1. Run the program and click the second icon on the main window.

2. On Capture Options window, select network interface of your PC.

3. On the same Capture Options window activate all Display Options.


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4. Again on the same Capture Options press Start button.

5. IP messages will appear on the main window:


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6. After recording the instance, stop the recording from “Capture” menu or stop icon in the mainwindow.

7. Save your recordings to a file from “File” menu or save icon in the main window and then send thatfile to Karel.


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VII.2. SIP LISTENER USE

SIP Listener is a module of KARELNET program, which is prepared for maintenance purposes. Itmonitors the signaling on the Karel IP modules (with SIP protocol). For the cases where Hub or programmable switch is not available for Ethereal logs, logs of even just the signaling part can providevaluable data.

The use of SIP Listener is really simple with few steps:

1. Open KARELNET and run its “SIP Listener” option.

2. Open Properties menu and enter the following options. Press “Save” and then exit the window.


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3. Type the IP address of the Karel module that you will monitor.

4. Press Start and check that the box “Conn.” has turned green to indicate the successful connection.Messages will appear on the left side of the window. These messages will be stored in the files thatautomatically form under the directory SIPLOGS. The files can be distinguished from date & IPaddress of the module. Send all of these files to Karel.


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VIII. SECURITY

When it comes to IP communication, to offer a secure communication becomes a very criticalrequirement. On user side (in other words end points like IP extensions or IP trunks), security

assurance is simpler: The end user can use sRTP protocol for speech and TLS for signaling for asecure communication

There are controls on Karel DS series systems that facilitate secure communication. These controlsare operational even in the absence of sRTP / TLS. Similar controls can be activated on the networkcomponents that lie outside Karel system.

This section explains these controls. If these controls are not taken into account, hackers can makeuse of the resources of your communication system out of your control which may be harmful.

CONTROLS ON NETWORKIf Internet access is available on a network, following steps must be controlled with networkadministrator of the installation site. (If a closed network like point-to-point VPN is used, these controlsare not required.)

Modems/routers that are used as Internet access points must have extra capabilities of atleast a simple firewall and must be able to handle IP port forwarding.

Static IP must be a pre-requisite for the parties that will be reaching the modem – firewall. Onthe system side, modem – firewall must be configured to accept requests coming only fromthese known static IP addresses.

The maintenance passwords of ADSL modems must be changed to a different password fromthe factory default. A password of at least 8 characters with at least 2 letters must be used.

80 th port of modems that can normally be managed from Web must be closed.

I I mm p p oo r r t t aa nn t t :: C C oo nn t t r r oo l l ss t t hh aa t t c c aa nn bb ee mm aa d d ee oo nn K K aa r r ee l l ss i i d d ee aa r r ee nn oo t t t t oo t t aa l l l l y y ee l l i i mm i i nn aa t t i i nn g g t t hh ee p p oo ss ss i i bb i i l l i i t t y y oo f f uu nn aa uu t t hh oo r r i i z z ee d d aa c c c c ee ss ss .. T T hh ee g g uu aa r r aa nn t t ee ee d d ss oo l l uu t t i i oo nn aa g g aa i i nn ss t t t t hh i i ss r r i i ss k k i i ss t t oo uu ss eeaa c c l l oo ss ee d d nn ee t t w w oo r r k k ss t t r r uu c c t t uu r r ee ( ( V V P P N N aa c c c c ee ss ss ) ) ..

CONTROLS ON SYSTEMTo disable unauthorized system access through IP network, following settings must be activated onIDEA:

PABX passwords must be defined from “Security Operations” menu of IDEA. Thus, each IDEAuser that will be connected to the system has to first enter the system’s password.

When LCR is used, LCR routes that are used to access PSTN lines must have access levelsgreater than 0.

When LCR is not used, the system must be configured to refuse external call requests comingfrom VoIP gateways or IP trunks.

(Whenever possible) If LCR is not used, line access of port 0 must be prohibited.

If the system is using a LAN adaptor card, IP addresses that can connect to your system for IDEA, KNEE or Net-CM programs must be determined through a local KNEE connection.

CONTROLS ON VOIP GATEWAY – IP TRUNKTo disable unauthorized system access through IP network, following settings must be activated onKNEE:

A KNEE of version AAQ or better must be used to activate password. A password of at least 8characters with at least 2 letters must be used.

“Apply IP Security” must be enabled from “Protocol Definitions” menu on KNEE. With thissetting, only the IP addresses that exist in target list of Address table can reach VoIP gateway


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or IP trunk. (This setting is available with AAT and AAS software versions of new and old typeVoIP gateway cards.)

It is recommended to change the default signaling port 5060 to another unused port value if possible throughout the whole network.


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DS Series IP Guide

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