hybrid ip pbx february 2014

Hearty Welcome!

ETERNITY as Hybrid IP-PBX

Agenda

Introduction

LAN/ WAN Port Configuration

Mac Cloning

Dynamic DNS

VoIP Server Domain

VoIP Port Parameters

SIP Extensions

SIP Trunk

SIP Hardware Template

Matrix Extended Phones

Introduction

Soft Phone

VoIP Phone

Mobile

Analog Phone

Internet

Introduction

Hybrid IP-PBX means PABX which supports IP Extensions and TDM/Analog Extensions

Hybrid IP-PBX can also have different type of trunks like CO, ISDN, GSM, etc. depends on hardware supported by IP-PBX

ETERNITY supports only SIP protocol for VoIP

Such system has capabilities to convert media between IP and TDM

SIP Resources

ETERNITY VARIANT SIP EXTENSIONS SIP TRUNKS VOIP CHANNELS/CARD

ETERNITY PE3SS

ETERNITY PE3SP

ETERNITY PE6SP

ETERNITY GE6S

ETERNITY GE12S

ETERNITY ME10S

50 16 16

500 16 32

1000 32 32

ETERNITY ME16S 1000 32 32

Agenda

Introduction

Mac Cloning

Dynamic DNS

VoIP Server Domain

SIP Extensions

SIP Trunk

LAN Port Configuration

Name can be assigned just for

identification

Hardware Slot & Port Offset

Customization is not possible

MAC Address of LAN Port

Configure IP Address and Subnet Mask for LAN Port

LAN Port doesn’t support DHCP connection

LAN Port Parameters

LAN Port is available in VoIP server card so all SIP extensions in local network with VoIP card can register without using WAN [Internet]

LAN PORT

WAN Port Configuration

MAC Address of WAN Port

Customization is not possible

Enable/Disable MAC Cloning using this flag

Configure Clone MAC Address

Agenda

Introduction

Mac Cloning

Dynamic DNS

VoIP Server Domain

SIP Extensions

SIP Trunk

What is MAC Cloning?

MAC cloning means to configure new MAC address [MAC-2] for the host without changing existing MAC address [MAC-1]

After doing MAC cloning host sends newly configured MAC address [MAC-2] in Ethernet Frames in place of sending

existing MAC address [MAC-1]

How MAC Cloning works ?

ISP Is Authenticating Host With MAC Address

Cloned MAC:- 01:1d:1a:02:82:34

Fix MAC:- 02:2d:1c:02:32:45

Why MAC Cloning?

Many times ISP tracks MAC address of host installed at customer premise to authenticate him as valid customer to provide Internet service

Due to this reason customer can access Internet only from single Host

Configuration of MAC Cloning

Select the internet Connection Type here Options: - Static

- PPPoE - DHCP

If the selected internet Connection Type is

‘PPPoE’,program the User ID, Password and PPPoE service

name here

If “Static” option is selected for DNS Address Assignment, then program the IP address of DNS and Domain

Name here

Select the DNS Address Assignment option here

(Auto/Static). If the selected option is ‘Auto’ then there is no need to

program the DNS address. It will be automatically assigned by the Service Provider/DHCP server

Agenda

Introduction

Mac Cloning

Dynamic DNS

VoIP Server Domain

SIP Extensions

SIP Trunk

What is Dynamic DNS?

Dynamic DNS means assigning a Domain Name to such host whose IP address changes frequently

Due to facility of DDNS that host can always be accessible from WAN by using same Domain Name

Why Dynamic DNS?

DHCP Connection

What is system’s

present IP?

1st attempt: 116.72.127.98

2nd attempt: 117.89.97.123

3rd attempt : 115.161.181.183

4th attempt: 118.187.24.89

Why Dynamic DNS?

In this case host will not be accessible always using public IP assigned to it by ISP

When ISP gives Internet connection type as PPPoE or DHCP then IP assigned to router at client site may be changed frequently

It can be resolved by using Dynamic DNS

How DDNS works?

DDNS Server is accessible globally, it keeps details of domain name

and global IP of all customers

DDNS Client Which Gives Update To Server

About Global IP Of Router

Matrixcomsec.dyndns.org : 203.88.143.221

Matrixcomsec.dyndns.org : 115.23.143.241

115.23.143.241 203.88.143.221

Internet Cloud

Dynamic DNS Configuration

Enable/Disable Dynamic DNS here. Enabling this option will help the VoIP

card to inform the SIP clients to pass the information of latest IP assigned to the VoIP card by the DHCP or PPPoE Server

Turn ON this option if the internet Connection type is DHCP or PPPoE

and DDNS option is enabled

DDNS option will be useful only if the Internet Connection Type is DHCP or PPPoE

Program the ‘Password’ provided by Dyndns.org

here, if the DDNS option is enabled

Program the ‘User-ID’

provided by Dyndns.org here, if the

DDNS option is enabled

Program the Host Name provided by Dyndns.org

here, if the DDNS option is enabled

Number of request send by the VoIP Card to DDNS

Server for the IP update request. Applicable only if

DDNS is enabled

This option helps the VoIP Card to re-establish the

mapping with the DDNS if the IP update request has not been sent in time by

the VoIP Card

It shows that VoIP card has successfully sent request to DDNS

server to update router’s public IP

detail in database of DDNS server

It shows that VoIP card failed to send request to DDNS server to update

router’s public IP

Check configuration Gateway and

DNS IP

Agenda

Introduction

Mac Cloning

Dynamic DNS

VoIP Server Domain

SIP Extensions

SIP Trunk

VoIP Server Domain

With this option when user will send SIP messages then VoIP card will listen for SIP message which is

redirected to programmed Domain Name and WAN IP Address

VoIP Server Domain

If client already have fix Domain name purchased from DNS service provider then that DNS can be configured here

That DNS will be assigned to VoIP server card

All SIP users from WAN can register to this DNS assigned to IP server card

Mostly public IP mapped to this Domain remains fixed that make it different from DDNS

VoIP Server Domain

Click on “Advance” to get detailed parameters

Agenda

Introduction

Mac Cloning

Dynamic DNS

VoIP Server Domain

SIP Extensions

SIP Trunk

Simple Traversal of UDP through NATs

UDP (User Datagram Protocol) is a Network Protocol for Transmission of Data

STUN allows VoIP Card to work behind Asymmetric NAT

STUN Client (VoIP Card) sends a request to STUN Server

Router

STUN Server

STUN Client

STUN Client requests STUN Server

Server updates with IP address used by router and open port to client

Client uses this information of IP address and free port from the server to ETERNITY NE

STUN will not work if the Router’s NAT Type is ‘Symmetric’

Illustration of STUN

Router with public IP STUN server

SIP server

Invite 203.88.142.119:5063

200 OK

ACK RTP

Select this options only if you have not forwarded the SIP & RTP

Listening Port in the Router. If flag is “Enabled” then system will use the SIP & RTP listening Port information

provided by the STUN Server

Program the STUN Server IP Address here

Program the STUN Server port here

STUN Configuration for SIP TRUNK and Extensions

STUN will be effective only when “Source Port IP Address” option is selected as “Use IP Address Fetched using STUN”

Source Port IP Address can be configured in “SIP Extension General Parameters” and in “SIP Trunk Parameters”

STUN Configuration for SIP Extensions

STUN Configuration for SIP TRUNK

Agenda

Introduction

Mac Cloning

Dynamic DNS

VoIP Server Domain

SIP Extensions

SIP Trunk

VLAN (Virtual LAN)

VLAN is good option for big

network to give high data speed

VLAN (Virtual LAN)

Priority can be defined to SIP packets on Layer2 level

Priority can be defined to RTP packets on Layer2 level

Agenda

Introduction

Mac Cloning

Dynamic DNS

VoIP Server Domain

SIP Extensions

SIP Trunk

VoIP Port Parameters - QoS

This field defines the priority Bit for all the SIP message sent

by VoIP card. Range 00-63

This field defines the priority bit for all the RTP message sent by VoIP card.

Range 00-63

Public IP

INTERNET

115.118.161.163

Users can directly access the device over

internet

(Public IP Address)

Router’s Public IP Address

Public IP Address of the NAT Router behind which VoIP card is installed. Program the Router’s IP only if the option of Router’s

Public IP is selected in ‘SIP Trunk Settings’

Router’s Public IP Address for SIP Trunk

Router’s Public IP Address for SIP Extension

If ETERNITY detects absence of RTP packets till expiry of this timer then it will disconnect the call

This much of channels will not be available for SIP extensions

Following number of physical channels reserved for SIP Trunks

Enable this flag, this will make the VoIP card to use ‘100rel’ extension along with all the SIP provisional

messages

100rel and SIP PRACK

SIP PRACK (SIP Provision Acknowledgement) is a method to enable reliability for SIP 1XX messages

The Called Party answer the PRACK by 200OK and PRACK is only for 1XX

messages other than 100 Trying

Generally PRACK message flows from Calling Party to Called Party

100rel and SIP PRACK

To get more reliability on SIP messages

Enabling this flag will make the VoIP card to send the SIP messages over TCP

SIP Listening and Source Port for UDP

Range 1025-65535

RTP Listening and Source Port Range 1025-65278

SIP Listening and Source Port for TCP

Range 1025-65535

SIP Listening and Source Port for TLS Range 1025-65535

This timer should be less then UDP binding timer in router

(Range 001-999 seconds)

Enable this flag to keep UDP binding refreshing in NAT router “Notify” or “Register” message can

be sent to keep UDP binding alive in router

This timer should be less then TCP binding timer in router (Range 0001-

9999 seconds)

Enable this flag to keep TCP binding refreshing in NAT router

This is the timer for which system waits for a response from the

called party after sending INVITE message. On expiry of this timer,

system terminates the call

This timer starts on the receipt of the provisional response

receipt from the called party and stops at the final receipt of

response. On this timer’s expiry, system disconnects the

This timer is applicable to all request, system will clear transaction after expiry of timer if it will not receive

response for sent request

LED2 on VoIP card will show status of SIP trunk defined here

(Range 01-32)

Agenda

Introduction

Mac Cloning

Dynamic DNS

VoIP Server Domain

SIP Extensions

SIP Trunk

What is SIP Extension?

Like any SLT and DKP, ETERNITY can have extensions that can be connected via internet/ LAN

ETERNITY VoIP Card can work as SIP Server to register SIP extensions from LAN, WAN or VPN

SIP Extensions

SIP Extensions Features

Hold Other Extension

Change User Status

Call Budget Toggle Two Calls

Publish/IM CUG

DND (Do Not Disturb)

Dial Operator Transfer Held Call Selective Port Access

Set message on (SLT/DKP)

Self Ring Test

Call Forward Alarm Reminder

Dial Floor Service Group

Room Monitor on Idle DKP Port

Use 3 Parties/ Dial In/Multi Party Conference

Personal/ Global Directory

Group Call Pick – Up

Voice Guided Alarm/ Reminder

Auto Call Back on Busy/Ringing Call

Use Walk – In Feature

Recall to last Caller

Use Keyboard Macro

Selective Call Pick – Up

Dial SA/SE Command

Park Other/SIP Extension

Voice Message Notification

Account Code

CLI Restriction

Forced Answer Feature

Retrieve Parked Call

Use Busy Lamp Field

Emergency Number

Features Other Extension can Use with SIP Extension

Set Call Forward on SIP Extension

Apply Raid on Busy on Busy SIP Extension

Park SIP Extension

Use Walk – In for SIP Extension

CO Call Waiting

Configure SIP Extension in Hotel/Enterprise Installation Wizard

Apply DND Override on SIP Extension

Hold SIP Extension

Retrieve Parked Call

Set Hotline on Extension

Call Supervision

Apply IR (Interrupt Request), BI (Barge-In)

Transfer Held SIP External Call

Selective port Access

Background Music

Hot Desk

SIP Extensions Settings

Configuring SIP Extensions

Server End Client End

SIP ID

Authentication ID

Authentication Password

SIP ID

Authentication ID

Registrar Server Address

SIP Extension Settings

Assign VoIP Software Port Number here

Use this flag to enable SIP extension

Configure Name of SIP extension user here, it will be displayed as

caller ID during internal calls (maximum of 18 characters)

If It is “Blank” Then called party will not get name received from INVITE As CLI

Configure SIP ID using which SIP extension user will register with registrar of VoIP card (it can be up to 6 digits, 0 to 9, * and # are valid

digits)

All extensions can call to SIP extension

user using this number

VoIP card’s registrar will use this ID to authenticate SIP user (it can be configured

up to 6 digits, 0 to 9 , * and # are valid digits)

It will not be applicable If All

“Authentication” options are disabled

VoIP card’s registrar will use this password to authenticate SIP user (it can

be configured up to 24 digits, 0 to 9, * and # are valid digits)

Default password:

SIP extension user can make/receive maximum this number of calls

simultaneously (Range 01-10)

By enabling these flags you can authenticate SIP users during

these different request messages

SIP Extension Authentication

ETERNITY VoIP card uses MD5 algorithm to authenticate SIP users by using Authentication ID and Password

During specific events ETERNITY VoIP card can authenticate users by asking them to send Authentication Id and Password configured in SIP user device

[SIP Phone]

Types of Authentication

ETERNITY VoIP card can Authenticate SIP user during following SIP messages

REGISTER Request

INVITE Request SUBSCRIBE

Request

Voice Mail subscription

BLF subscription

Presence Subscription

By enabling this flag you can get notification on call states of al the

phones with the same SIP ID at different locations

Enable this option to get Voice Mail Notification on

VoIP phone

VoIP phone should support Voice Mail Notification

feature

To allow this SIP extension user to view the status of the

availability of other SIP enabled terminals, this flag should be

enable

BLF Key in SPARSH VP248

LED Glowing RED: User Busy LED Blinking RED: User Ringing

By enabling this flag VoIP Phone users can publish their availability status

By enabling it VoIP server will ask for

authentication details from SIP users when

receives PUBLISH message

By enabling this flag VoIP Phone users can see availability status of

other SIP/TDM users

Other VoIP/TDM users should publish

their availability status to use this

feature

Soft phone user 3304 is subscribing status of 3301

and 3303 (All SIP users registered with

VoIP Server Card)

It is obvious that 3301 and 3303 are publishing their availability status to VoIP

Server Card

It completely depends on SIP user that which type of availability status it

can Publish or Subscribe

Publish Status of DKP & SLT

DKP and SLT users can also publish their availability status by applying simple commands from their phones

Following is sequence to dial

commands

Off Hook SLT/DKP Phone

Dial 104

Feature Tone---User Password

Enter code [Range 0 to 9]

Publish Status of DKP & SLT

Code Status

0 Absent

1 Present

2 Auto Detect

3 Away

4 On the Phone

5 Do Not Disturb

6 I am on Mobile

7 In Meeting

8 Out for Meal

9 Out of Office

Publish Status on Soft Phone

Different SIP hardware parameters can be assigned

to different SIP users

Same like SLT and DKP users following Features can be assigned to SIP users also

Same like SLT and DKP users Call Pick Up group can be assigned to SIP users also

If system is configured to use in hotel mode then SIP extension can

also be configured as “Guest”

SIP Extension General Parameters

This is Name which you have assigned to VoIP server card

Showing Hardware Slot and Port of VoIP card

Select here which IP should be considered as source IP when VoIP card communicates with

SIP users

VoIP card will receive registration request from SIP users only between this timer

interval

Registration Timer Configured in SIP

users must be between this values

Following Private Key is used to encrypt SIP message

[MD5 Authentication] It can be up to any 24 ASCII character

Agenda

Introduction

Mac Cloning

Dynamic DNS

VoIP Server Domain

SIP Extensions

SIP Trunk

SIP Trunk Configuration

VoIP calls can be initiated after suitable programming of SIP Trunks

ETERNITY supports 2 types of SIP trunks: Peer to Peer and Proxy

Peer-to-Peer Calling

203.88.143.218 204.88.142.218

Internet TCP/IP

Making a VoIP call directly to the destination without any intervention of any mediator is called

peer-to-peer calling.

You just need to know the called party’s IP address.

Select Peer-To-Peer in the SIP Trunk Mode You can select either

Trunk or Station

If you select Trunk, then it will follow the

Trunk Feature Template as per the

SIP trunk

Configuration: Peer-to-Peer Calling

If you select Station, then it will follow the direct landing on specified

extension

Calling 205

201 205 207

Proxy Calling

Making VoIP calls through proxy server is called proxy calling

Proxy Server: abc.com

Client 2 SIP ID 402

Client 3 SIP ID 403

401 calling 402

Client 1 SIP ID 401

Requirement for Proxy Calling

Proxy server authenticates the clients for outgoing calls through it

What is required for

authentication?

SIP ID

Authentication ID

Registrar Server Address

Registrar Server port

Proxy Calling

Program SIP ID here as per given by ITSP Program Registrar Server

address here. To be obtained from ITSP

Program Registrar Server port here. To be obtained from ITSP

(1025-65535)

Proxy Calling

It is the timer after which

request has been sent again

Registration retry if registration request is not acknowledged

User ID & password given by ITSP for authentication

Proxy Calling

Enable outbound proxy from here

Enter outbound proxy sever address here

Enter outbound proxy sever port here

Proxy Calling

Define SIP hardware

template here

Define TFT here if the SIP Trunk entity is “SIP

Client”

SIP Trunk Properties

Define Cost Factor here

Used in Gateway Application (01-64)

Enable RCOC here

SBFT, SAFT are applicable on the SIP Trunk if the SIP Trunk entity is P2P

By enabling these flags you can authenticate SIP user during

these different request messages

Enable if you want to send CLI on SIP trunk

Enable if you want to accept IC calls without

Define Source port IP address here

Enable/disable Digest Authentication

Enable Symmetric RTP from here

Why Symmetric RTP?

Symmetric RTP can be used in firewalls, debugging and troubleshooting.

Generally it is useful to resolve bidirectional speech problems.

Many firewalls, NATs, RTP implementations don’t work on asymmetric RTP but require symmetric RTP.

Digest Authentication

It is a security feature which is used by VoIP card during peer to peer incoming call

On any incoming SIP call, VoIP card will check the authenticity of the SIP user by Authentication ID and Password

This authentication is done by using Digest Authentication Table

If authentication doesn’t match, VoIP card will reject the incoming SIP call

SIP users configured with following User ID and Password will only be allowed to access ETERNITY during

Peer To Peer calling

Enable this flag in SIP Trunk

parameters

Select default transport for outgoing messages i.e. UDP, TCP

or TLS

UDP v/s TCP v/s TLS

UDP TCP TLS

UDP is connectionless and acknowledgement less protocol (DNS, VOIP)

TCP is connection oriented & provides acknowledgement (WWW, FTP, E-mail)

TLS is connection oriented protocol & provides acknowledgement

Used for time sensitive applications

TCP requires more bandwidth than UDP

TLS requires more bandwidth than TCP

Used for servers that answer small queries from huge number f clients

Used in the applications where secure connection is required & data loss should be less

When secure transportation is to be used

Define IC ref. ID & OG ref. ID here for DDI mapping

Program the value of

pause timer here (1-9)

Used during Gateway Application

This is set as per the requirement of

remote peer

Enable it when SIP trunk is to be generated for an

invite without SDP

Agenda

Introduction

Mac Cloning

Dynamic DNS

VoIP Server Domain

SIP Extensions

SIP Trunk

Select Preference for Vocoders according to compatibility of SIP users and expected VoIP

quality

Bandwidth requirement for each Vocoder is different

When G.723 negotiated then selected Bit Rate will be applied to send RTP (5.3 or 6.3 Kbps)

By enabling this flag Silent RTP packets will not be sent during conversation

Used for efficient usage of available

bandwidth

TX and RX speech level can be changed from here

Select DTMF type from following options

- RTP (RFC 2833) - SIP Info - In band

Same DTMF option must be configured in SIP user device

If RTP(RFC 2833) is

selected then this should be

configured

By enabling this flag Echo Cancellation will be activated when SIP users are talking to

Stations/Trunks [Analog/Digital]

This parameters can be configured according to strength of Echo required to

be cancelled

Separate options available for analog

and digital interfaces

Jitter Buffer

By considering packet network jitter can be defined as variation of delay in receiving packets

To resolve this problem the mechanism used in VoIP device is called “Jitter Buffer”

VoIP device stores packets according to Jitter Buffer timer supported by it to maintain common delay between successive packets before processing them

for regeneration of voice

Select “Static” option for network having precise delay but when delay is

not fixed always from network side then select option “Dynamic” Configure Jitter Buffer

timer here

Configure it for Dynamic Jitter Buffer

option

Select Protocol for FAX over IP. Following are options:

- T.38 (UDPTL) - T.38 (RTP)

- Pass Through

FAX Parameters is customized when protocol is selected as T.38

FAX Parameters customization when protocol selected as Pass Through

White List IP Address

ETERNITY supports security on Transport Layer

By enabling this security option ETERNITY VoIP card will accept incoming

traffic only from those VoIP devices whose IP address is configured in White List Table

White List IP Address

Configure here, IP Addresses of devices from where you want to receive incoming call traffic on VoIP card

Enable this flag to use IP level security

Static Routing

Some times customer have multiple routers in their network to connect their multiple sites using MPLS (Multiprotocol Label Switching)/ Frame Relay

In same network there can be distinct routers to connect to Internet

In such network scenario to connect VoIP devices at multiple sites using point to point connectivity there is need of some mechanism to route the calls from different

router according to IP address of different destination VoIP devices

Static Routing

Agenda

Introduction

Mac Cloning

Dynamic DNS

VoIP Server Domain

SIP Extensions

SIP Trunk

Matrix Extended IP Phone

SIP extensions we registered just previously are called as Open SIP Phones. These phones do not work as a DKP.

Matrix provides its proprietary IP phones to register as an Extended IP Phone which will work as it is as a DKP

Matrix SPARSH VP248

Matrix SPARSH MS

Programming Steps-Eternity

Program VoIP Port No., Name, SIP ID, Authentication ID,

ID Authentication

Password for SIP

In the Location menu Enable Matrix

Extended Phone mode and define MAC address of SPARSH VP248

Note: User can register Matrix Extended IP phones at three different locations, i.e. a single account can be registered on three IP phones

Configure the Master CPU IP address

This port is to be assigned in the VP

phone where server port is

needed

Matrix Extended IP Phone- SPARSH VP248

Server port is 80

Master CPU IP Address

Matrix Extended IP Phone- SPARSH VP248

Matrix Extended IP Phone- SPARSH MS

SPARSH MS is a mobile softphone client for android smartphones and iPhones for consistent in-office experience.

You can use Wi-Fi or cellular networks to connect to the system while working from office, home or travelling to any location.

There is a flexibility to reach to office users with direct extension number dialing.

Download Matrix SPARSH MS from play store if you have android device or from apple store if you have iPhone.

Configuring Matrix Extended IP Phone

Video Calling through VoIP

Matrix provides video calling facility also with VoIP calling only.

The phones we use for example

Video Calling with SPARSH M2S

Download Matrix SPARSH M2S from play store if you have android device or from apple store if you have iPhone.

All the settings of SPARSH M2S is similar to SPARSH MS for the server settings at ETERNITY server end

After you install the application of SPARSH M2S; let us suppose you are using an android tablet for video calling

You will have to enter the credentials as per

the server settings (Extended phone type

settings)

User is registered properly

XYZ is calling another SIP

extension 615

Option to start video

Option to start audio

Option to send IM

Option to make a video call

Here it will be your video

During an audio call, you can

switch it over a video call by selecting this

option

Options: Video Calling with SPARSH M2S

Video Calling with Bria Soft phone

Option to start video

Enter the credentials in the SIP account of

Bria phone

Video Calling with Bria Soft phone

Video Calling with Linphone

Enter the SIP extension number

whom you want to call

Video calling: Caller’s video

will come

hybrid ip pbx february 2014

existing mac address

configured mac address

new mac address mac

configuration of mac

ip address of dns

mac cloning host

lan port configuration

clone mac address

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