TRANSipVoice over IP solution for REDCOM Technologies

Corporal Neo Martinez Communications CompanySeptember 2013

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Overview• The purpose of this document is to inform you, the user, on

fundamental TRANSip concepts and configurations to provide VoIP capabilities with REDCOM Technologies.• In other words (Marines): instead of setting up a Call Manager to

install voip phones, you will now program TRANSip in MSC Slot 15 of your Redcom.

Note: the audience targeted for this lesson are Marines with no prior experience of IP telephony.

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Overview• The areas that will be covered during this lesson are:

• Media Service Circuit (MSC) Board• TRANSip Capabilities • Configuring internal VoIP phones with SIP signaling • Configuring a SIP to trunk to external nodes

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Media Service Circuit (MSC) Board

• The MSC Board provides Dual Tone Multi Frequency (DTMF) detection, tone generation, echo cancellation and conference calling.

• The MSC Board is featured in the Slice2100 and the High-Density Exchange Commercial Communication Switch (CCS) inside the DTC facility.

• The MSC Board in compatible with the Modular Switching Units (USMC’s DEOS), however, it must run REDCOM v5.0 or higher.

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Media Service Circuit (MSC) Board• Located in slot 15 of USMC packages.

HaltMajorPower Good

N/A

N/A

RGCP Registration DSP 0

DSP 1

DSP 2

DSP 3

DSP 4

DSP 5

100Base TX10Base T

TX

Link

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RJ-45 Connector

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TRANSip• TRANSip is Redcom’s solution to IP telephony.• TRANSip uses Session Initiation Protocol (SIP) protocol to

communicate between devices.• You will need to Configure a LAN Switch for network

connectivity.• Consult a Data Marine or network administrator for further

guidance on setting up a LAN Switch.

NOTE: TRANSip uses SIP to communicate its devices, that means you will need to modify the voip phone’s firmware. For further guidance on how to change firmware visit: http://www.neoathome.com/sip-firmware-load/

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Elements of TRANSip • TRANSip Consist of 4 fundamental elements;

• SIP Call Manager: provides functions such as call setup, control, termination, and directory services for IP users.

• Media Gateway: supports various compression algorithms/CODECS such as FoIP, G.711 and G.729.

• Media Gateway Controller: TRANSip acts as a gateway to allow different media/protocols to communicate. (I.e. VoIP, BRI, PRI, DTMF, ect. )

• Legacy Support: this feature makes TRANSip compatible with external T!, E1, E&M and LSRD.

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Configuring TRANSip• Next you will configure TRANSip to make internal calls with

voip phones.• PLEASE NOTE:

• Before you begin you need to set a LAN Switch for network connectivity.

• You MUST change the voip phone’s firmware to SIP rather than SCCP. Click on the hyperlink and follow this firmware tutorial If you need guidance on how to do this protocol change for your voip phones.

• Assuming you have basic POTS internal calling and you met the requirements mentioned above, you may begin configuring the REDCOM for voip purposes.

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Configuring TRANSip• Follow the commands and understand what you’re typing to the console.

Refer to the snap shots in the following slides for a visual reference/guidance.

Ethernet Configurationadm>slot=p enter network settings see image 1adm/slot>ip address 192.168.1.2 assign the MSU IP addressadm/slot>net_mask 255.255.255.0adm/slot>gateway 192.168.1.1adm/slot>ex;ac

Assign the MSC Board an IP Addressadm>slot=15 enter MSC Board network settings see image 2adm/slot> ip address 192.168.1.5 assign MSC IP (or call manager IP)adm/slot>net_mask 255.255.255.0adm/slot>gateway 192.168.1.1adm/slot>ex;ac

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Configuring TRANSipDevice Screen | Configuring voip phones

adm>device enter device settings see image 3adm/device>entry=22adm/device>state=on turn sip service on adm/device>link=23adm/device>entry=23adm/device>state=on adm/device>port=5060 layer 4 port number for sipadm/device>ex;ac

Configuring Line Groupadm>group create a line group for internal voip phones see image 4adm/group>new=linadm/group>group=1 note:your first groupadm/group>type=voip set protocol to voipadm/group>name=“IP-hone”adm/group>members see image 5

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Configuring TRANSip

adm/group>add=15/0/0 add MSC circuit board to your voip line groupadm/group>qty=15 number of devices adm/group>ex;ac

Setting Dynamic Station List Attributesadm>dtsnlist>station=100 create a new station see image 6adm/dtsnlist>name=Majoradm/dtsnlist>dtsn=100 access the voip station you just madeadm/dtsnlist>signaling=sip set parameters see image 7adm/dtsnlist>voip_group=1 should match you voip line groupadm/dtsnlist>sip_user=100adm/dtsnlist>ex;ac

Secure Session Initiation Protocol (SECSIP)adm>optionadm/opt>secsip set secure sip see image 8adm/opt>ex;ac 11

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Configuring TRANSipIncreasing the DSTN table for more users

>gengen>data this allows more space for voip users, default is only 2gen/data>table=dstint see image 9gen/data>current=20 set user value to the number you wish gen/data>ex;ac

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Setting up a SIP TRUNK• A SIP Trunk will allow you to talk to other nodes over IP. Configuring a SIP Trunk

adm>groupadm/grp>new=trk create a second group for your sip trunkadm/grp>group=2adm/grp>name=“SIPtrunk”adm/grp>dialing=sip set protocol to sipadm/grp>dct=11 route it out dial code table 11adm/grp>d_host_sip=192.168.1.5 your own MSC IP is hostadm/grp>d_name_sip=“sip”adm/grp>membersadm/grp>add=15/0/0 add the MSC circuit board adm/grp>qty=24adm/grp>ex;ac

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Setting up a SIP TRUNKConfiguring route screen

adm>route=1 create a route for your sip trunk see image 10adm/rte>name=“sip to xxx”adm/rte>group=2 assign it to match the sip trunkadm/rte>del=0adm/rte>out=7adm/rte>ex;ac

Configuring Dial Code Tablesadm>dctadm/dct>dct=7adm/dct>entry=9;pattern=xxxxadm/dct>type=rte;val=2adm/dct>dct=8adm/dct>entry=9;pattern=nnxxadm/dct>ex;ac 14

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