Technical Training
IP & VoIP fundamentals
IP based Products overview
VOIP Client & server Capabilities
VoIP
IP Basics
The TCP / IP Model
Internet Protocol, IP is
an address of a computer
or other network device
on a network using IP or
TCP/IP
IP Ranges of Different Classes
Internet Protocol Version 4 : 4 Octets
4,294,467,295 IP Addresses
IPv4 E.g. : 11010001.11011100.11001001.01110001
Decimal : 209.156.201.113
Internet Protocol Version 6 : 16 Octets
3.4 * 10^36 IP Addresses
E.g. : 11010001.11011100.11001001.01110001.
11010001.11011100.110011001.01110001.11010001
IPv6 .11011100.11001001.01110001.11010001.11011100.
11001001.01110001
Decimal :
A524:72D3:2C80:DD02:00029:EC7A:002B:EA73
IPv4 Ranges of Different Classes
1.0.0.1 to 126.255.255.254
Class A
Supports 16 million hosts on each of 127 networks
128.1.0.1 to 191.255.255.254
Class B
Supports 65,000 hosts on each of 16,000 networks
192.0.1.1 to 223.255.254.254
Class C
Supports 254 hosts on each of 2 million networks
224.0.0.0 to 239.255.255.255
Class D
Reserved for multicast groups
240.0.0.0 to 254.255.255.254
Class E
Reserved for future use, Research, Development Purposes
Private IPv4 Address Range
10.0.0.0 to 10.255.255.255
Class A
Subnet : 255.0.0.0
2^8 Networks & 2^24 Hosts
172.16.0.0 to 172.31.255.255
Class B
Subnet : 255.255.0.0
2^16 Networks & 2^16 Hosts
192.168.0.0 to 192.168.255.255
Class C
Subnet : 255.255.255.0
2^24 Networks & 2^8 Hosts
VoIP
(Voice Over Internet Protocol)
What is VoIP ?
“VoIP (Voice Over IP) is a technology that enables one to make
and receive phone calls through the Internet instead of using the
traditional analog PSTN (Public Switched Telephone Network)
lines “
• VoIP Stands for Voice-over-Internet-Protocol
• It is a Technology Designed to deliver Voice Information using
Internet Protocol (IP)
• It is a combination of Hardware and Software to make
telephone calls via the Internet
• Voice Signals are Converted in to Packets of Data which are
Transmitted on Internet
VoIP Devices
IP Phone
• The phone able to connect itself directly to the internet for
VoIP communication
ATA
• Connects a standard phone to Internet for VoIP
communication
• The ATA is an Analog-Digital-Packet converter
IP Card for PBX
• Card with multiple channels for VoIP communication
VoIP Devices
Soft Switch
• PC Based soft IP PBX with PCI based Hardware for
PSTN interfaces
Soft IP Phone
• PC based Soft IP phones
Mobile Phone
• Mobile Phones with VoIP client software
SIP
(Session Initiation Protocol)
What is SIP ?
“ The Session Initiation Protocol (SIP) is an application-layer /
control (signaling) protocol for creating, modifying and
terminating sessions with one or more participants”
SIP can use either TCP or UDP for transport
RTP( Real Time Transport Protocol) is used in SIP for real media
transfer (Voice , Video etc.) . RTP is an application protocol that
uses UDP for transport
SIP V2.0 Protocol is the Protocol used by all MATRIX VoIP
Products
SIP Methods
INVITE Indicates a client is being invited to participate in a call session.
ACK Confirms that the client has received a final response to an INVITE request.
BYE Terminates a call and can be sent by either the caller or the callee.
CANCEL Cancels any pending request.
OPTIONS Queries the capabilities of servers.
REGISTER Registers the address listed in the to header field with a SIP server.
PRACK Provisional acknowledgement.
SUBSCRIBE Subscribes for an Event of Notification from the Notifier.
NOTIFY Notify the subscriber of a new Event.
PUBLISH publishes an event to the Server.
INFO Sends mid-session information that does not modify the session state.
REFER Asks recipient to issue SIP request (call transfer.)
MESSAGE Transports instant messages using SIP.
UPDATE Modifies the state of a session without changing the state of the dialog.
SIP Responses
1XX Provisional 100 Trying
2XX Successful 200 OK
3XX Redirection 302 Moved Temporarily
4XX Client Error 404 Not Found
5XX Server Error 504 Server Time-out
6XX Global Failure 603 Decline
VoIP Channel
What is a VoIP channel ?
“ Number of VoIP channels indicated the total number of
Simultaneous VoIP calls that can be made using a Particular SIP
Device”
For E.g. : In ETERNITY IP-PBX using a VoIP16 card we get a
total of 16 VoIP channels to make the VoIP calls and a VoIP32
Card Provides 32 VoIP channels
Same way SETU VP248 & SETU ATA range provide 2 VoIP
Channels i.e. at a time we can make two VoIP calls
SIP Trunk
What is a SIP Trunk ?
SIP Trunks may be Proxy or Non-Proxy
VoIP calls can be Initiated after suitable programming
of SIP Trunk number in the OG Trunk Bundle Group
A SIP Trunk can be configured either for Peer to Peer
calling or Proxy calling
Incase of ETERNITY VoIP Server card, a SIP Trunk
must be assigned to a VoIP Ethernet Port
Difference between
VoIP Channels & SIP Trunks???
VoIP calls can be done by using any of the SIP Trunk of
a SIP Device, However the total number of such
simultaneous calls that can be done for that device
depends on the VoIP channels provided
Total SIP Trunks associated with the device is also
known as the total number SIP Accounts supported by it
(When configuring for Proxy calling)
Client Application : Types of VoIP
Calling possible using SIP Trunk
Types of VoIP calling
A SIP Trunk can be configured either for Peer to Peer
calling or Proxy calling
In case of Peer to Peer calling the SIP Trunk has to be
configured as a Peer to Peer Trunk.
In case of Proxy calling, the Account details such as ID,
password & Server IP Address that are obtained from
SIP Proxy Server has to be configured for the SIP Trunk
Peer to Peer Calling
Peer to Peer Calling
Making a call on the VoIP Ethernet Port without going
through any proxy is called Peer to Peer Calling
In this case the User grabs the Trunk that is configured
for peer to peer calling and directly dials the IP or Short
code programmed for that IP in the Peer to Peer Table
Configuring: Program VoIP Ethernet Port, Enable the
SIP Trunk and configure the SIP ID as „*‟ for station
functionalities or configure any number if you require it
to have Trunk capabilities
Peer to Peer Calling
Artificially simple case: Caller knows where my phone is
INVITE sip:220.225.50.115
100 Trying - 180 Ringing - 200 OK
ACK
Caller’s
Phone 102 RTP session (voice, video, etc)
My
203.88.142.219 Phone 402
220.225.50.115
SIP Trunk : Peer to Peer in ETERNITY
P2P Call : Both Devices are in Public IP
Internet 203.88.142.218
Public IP
203.88.143.75
Public IP
P2P Call One Device is on Public IP and
Other Device installed behind NAT
Router separates
Port Forward in Private and Public
Router Network
203.88.142.218
Internet Public IP
LAN WAN
192.168.1.254 203.88.142.220
IP: 192.168.1.2
G/W : 192.168.1.254
Private IP
Peer to Peer: Call Flow
SIP (102@203.88.142.219) SIP (402@220.225.50.115)
INVITE SDP (402@220.225.50.115)
100 Trying
180 Ringing
200 OK
ACK
Media Session (RTP)
BYE
200 OK for BYE
Proxy Calling
SIP Proxy Calling
For Proxy Calling apart from ISP connectivity caller needs to buy
Proxy services i.e. a SIP Account from an ITSP (Internet
Telephony Service Provider)
The ITSP will Provide you with a SIP ID, Authentication Id,
Authentication Password and the Registrar Server Address /
Domain Name i.e. the ITSP‟s own address where the SIP Trunk is
to be registered
Configure the details obtained from the ITSP for the SIP Trunk in
the SIP Trunk Parameters
SIP Calling - Proxy
SIP Id : 403 SIP Id : 402
User Agent 3 User Agent 2
IP-Based
Network
SIP Id : 401
Proxy Server
User Agent 1
SIP Trunk No. 2 of
ETERNITY registered
abc.com
with Proxy server
abc.com
SIP Trunk : Proxy
Proxy Calling : Call Flow
SIP Agent (456@pqr.com) SIP Server (abc.com) SIP Agent (123@xyz.com)
INVITE SDP (123@xyz.com)
407 Proxy Authentication
required ACK
100 Trying
INVITE SDP (123@xyz.com)
100 Trying
180 Ringing
180 Ringing
200 OK (456@pqr.com)
200 OK 456@pqr.com
ACK
ACK
Authenticated Media Session (RTP)
BYE
200 OK for BYE
Server Capabilities : Registering of SIP
Extensions
What is a SIP Extension?
ETERNITY ME/GE/PE VoIP Server Card, ETERNITY NE VoIP
Server Module & SAPEX IP PBX Server have Server Capabilities
Thus in this Case they themselves behave as a Proxy Server and
Provide SIP Accounts to Other SIP Devices
The SIP device that gets registered to the Server thus become a SIP
Extension / User of the IP PBX and can avail the System resource
as well as make calls to other such user
Configuring SIP Extension
The SIP device (SIP Client) can be registered to the local IP of
Server incase it is in the same network as the Server else it can
Register itself using the Public Internet N/W provided the Server
as well as the SIP device are getting Internet
In the Server‟s User Settings or SIP Extension Settings first of all
we configure the SIP ID, Authentication ID & Password
Then these information i.e. the SIP ID, Auth ID, Auth Password
and Registrar server address are configured for the SIP Trunk of
the Client Device.
Configuring SIP Extension
As soon as the Client Device configures the Account details for its
Trunk, request is send to the Server to enable it to get registered
with it
The Server Authenticates the Client Device by verifying the
Authentication ID and Password & on Successful validation the
SIP Device gets registered and becomes a SIP Extension of the
Server System
SIP Extension Settings : ETERNITY
Proxy Calling : Call Flow
3302@203.88.142.221
Registered SIP Extension
3302
Ext. Mobile of ETERNITY
Dials
3001
VoIP
VoIP Server
Card IP :
203.88.142.221
VoIP
Dials 0 3001 3002 Registered SIP Extension
Grabs GSM of ETERNITY :
Trunk & Dials ETERNITY first
Ext. Mobile no. authenticates the SIP
Device with ID &
3301@203.88.142.221 Password only then it
Registered SIP Extension becomes its SIP Extension
3301
of ETERNITY
Some Concerns related to
SIP Extension and VoIP Channels
SIP Extension, SIP Trunk and VoIP
channels
When SIP Extension makes a call two another SIP Extension a
total of Two VoIP Channels is consumed
When a SIP Extension makes a call Using any other Trunk except
SIP Trunk of the System only one channel is consumed
When a SIP Extension makes a OG call using a SIP Trunk of the
System Two VoIP channels are consumed
When a normal DKP/SLT Extension of a system make OG call
using SIP Trunk 1 VoIP channel is consumed
Range of MATRIX Products
with VoIP Interface
ETERNITY ME / GE / PE VoIP
Interface : VoIP Server Card
ME ME GE GE GE PE PE PE
Hardware
10S 16S 3S 6S 12S 3SS 3SP 6SP
Maximum VoIP
Channels / 32 32 32 32 32 16 16 16
VOIP calls per
card
VoIP/ SIP Trunks 32 32 16 16 16 4 4 4
SIP Extensions 500 500 500 500 500 50 50 50
ETERNITY NE :
Hybrid IP PBX with Server Capabilities
8 VoIP Channels
4 SIP Trunks
Up to 16 IP Extensions
2 DKP Port
Up to 2 GSM Ports
2/3/4/6 FXS Ports
4/6/10/14 FXO Ports
SAPEX : Pure IP-PBX Server
Up to 500 IP Extensions
10 SIP Trunks
VoIP Channels depends on the
Transcoding and Codec Used
MATRIX IP Phones
SETU VP248S/SE SETU VP248P/PE
2 VoIP Channels 2 VoIP Channels
3 SIP Trunks 3 SIP Trunks
ATAs ( Analog Terminal Adaptors)
SETU ATA 1S SETU ATA 2S
1 FXS 2 FXS
2 VoIP Channels 2 VoIP Channels
3SIP Trunks 3 SIP Trunks
SETU ATA 211 SETU ATA 211G
1 FXO 1 FXS
1 FXS 1 GSM
2 VoIP Channels 2 VoIP Channels
3 SIP Trunks 3 SIP Trunks
MATRIX VoIP Gateways
SETU VFXTH
VoIP FXO FXS
Configuration
Channels Ports Ports
SETU VFXTH0016 16 0 16
SETU VFXTH0024 24 0 24
SETU VFXTH0032 32 0 32
SETU VFX 44L/VFX 88L SETU VFXTH0800 08 08 0
4/8 VoIP Channels SETU VFXTH1600 16 16 0
4/8 FXS Port SETU VFXTH2400 24 24 0
1 Life Line Port SETU VFXTH3200 32 32 0
9 SIP Trunks SETU VFXTH0808 16 08 08
SETU VFXTH1212 24 12 12
SETU VFXTH1616 32 16 16
32 SIP Trunks
MATRIX VoIP Gateways
SETU VGFX8422
SETU VGFX8404 SETU VTEP321
SETU VGFX8440 VoIP to T1/E1/PRI Gateway
VoIP-GSM-FXO-FXS Up to 32 VoIP Channels
8 VoIP Channels 32 SIP Trunks
9 SIP Trunks 1 T1/E1 PRI Port
4-GSM Ports Network Clock Synchronization
2/4-FXO Ports
2/4-FXS Ports
MATRIX VoIP Gateways
SETU VBR42
SETU VGB842
VoIP to ISDN BRI Gateway
VoIP- GSM-ISDN BRI Gateway
Plug-n-Play Configuration 4 VoIP Channels
8 VoIP Channels 2 ISDN BRI Ports
4 GSM Channel 2 Ethernet Ports
2 ISDN BRI Port Network Clock Synchronization
Network Clock Synchronization
VoIP Port Configuration in ETERNITY
VoIP Server Card : WAN & LAN Port
When the VoIP card is installed in a Public IP Network
- WAN Port of the card is connected to a Broadband
Router / Modem
- Public IP is assigned to the WAN Port
- LAN port is connected to a switch/hub to which SIP
devices are connected
When VoIP card is installed in a Private N/W, behind a NAT
Router
- WAN Port connected to the LAN Switch / Hub
- Private IP is assigned to the WAN Port
- SIP devices within the LAN can get registered with the
Card
VoIP Parameters in ETERNITY Jeeves
VoIP Port Parameters
Configure the LAN Port Parameters when
connecting the VoIP card in a Local Area Network
Using a Switch
VoIP Port Parameters
MAC Cloning enables to Clone the Unique
MAC Address of WAN Port to desired
MAC Address.
Configure the Cloned MAC Address if the
system needs to use MAC Address other
than the Pre-programmed one
VoIP Port Parameters
When Connection
Type : Static
Configure the 32 bit
IP Address, Subnet
Mask & Gateway
Address as provided
by the network
administrator or ISP
VoIP Port Parameters
DHCP: When you have DHCP server in
your Network the IP and other details
will be automatically assigned
(Parameters assigned by DHCP Server
displayed on „Network Status‟ Page)
VoIP Port Parameters
PPPoE: Select PPPoE when ISP provides internet
services on Point to Point Protocol over
ETHERNET.
Enter the : PPPoE User ID (Max 40 Characters)
PPPoE Password (Max 24 Characters)
PPPoE Service Name (Max 24 Characters)
VoIP Port Parameters : DNS
Configure the DNS Parameters here.
DNS (Domain Name Server) is used to resolve domain
names to IP Addresses.
Select „Yes‟ if DNS address is to be assigned automatically
by the DHCP/ PPPoE Server.
If „No‟ is Selected for Static/DHCP/ PPPoE connection then
configure manually the Primary DNS Address, Alternate
DNS Address & DNS Domain name as provided by the ISP
or N/W Administrator
VoIP Port Parameters : DynDNS
Configure the DynDNS Parameters whenever you
need to resolve the Internet Connection with a
Domain Name . IF the Connection type is DHCP or
PPPoE, IP Address doesn‟t remain static and
changes periodically, thus by assigning a DynDNS
name to the connection the DynDNS server
regularly updates the IP address associated with
the Domain Name and hence it will be possible to
make Peer to Peer call to the SIP Trunk even if the
connection is DHCP by using the Domain name
Steps to configure the DynDNS
Enter the Username &
Password. Create an account if
Open the DynDNS you are accessing it for the 1st
Server on dyndns.org time
*‘DYN DNS’ is trademark of Dynamic Network Services INC, USA
Go to My Services -
Add Host Name
Configure the
parameters for a new
Host Name
Select VoIP &
Add to Cart
The dyndns host will be
created. Click on Next
Total Hosts created
VoIP Port Parameters: Dynamic DNS
For Enabling Dynamic DNS configure the
parameters:- User Id (Max 40 characters), Password
(Max 24 characters) & Hostname(Max 40
Characters) as configured in the DynDNS server
STUN (Simple Traversal of UDPs
through NATs)
When the VoIP port (WAN) is located behind a NAT
Router & SIP Messages need to forwarded to the Public
Internet
STUN specifies the mechanism required for NAT
traversal in SIP messages. STUN server facilitates
traversing through most NATs except symmetric NATs
Illustration of STUN
STUN Request IP
STUN Request
Source:192.168.50.161:5060
Device
Source: 115.118.161.163:5060
With
STUN Response inbuilt
STUN Response
To:115.118.161.163:5060
To: 192.168.50.161:5060
STUN
Payload:115.118.161.163:5060
Payload:115.118.161.163:5060 client
STUN Server
VoIP Port Parameters: STUN
Program the STUN Server Address; Listening Port of
STUN Server (1024-65535) Default port : 03478; Enable
the Flag „Use SIP Port fetched using STUN‟ if SIP port
required to be fetched by STUN else disable when Port
Forwarding in the Router is done for SIP messages
VoIP Port Parameters:
Router‟s Public IP Address
Port Forwarding :
Since STUN doesn‟t work with symmetric NAT , as an
alternative to STUN Port Forwarding can be done in the
router and Router‟s Public address that is configured
can be used as Source Port IP Address
VoIP Port Parameters:
Router‟s Public IP Address
Program the Static Router‟s Public IP Address here
P2P Call One Device is on Public IP and
Other Device installed behind NAT
Router separates
Port Forward in Private and Public
Router Network
203.88.142.218
Internet Public IP
LAN WAN
192.168.200.210 203.88.142.221
SETU ATA
IP: 192.168.200.195
G/W : 192.168.200.210
Private IP
Router Configuration : Example
Router‟s
Network
Parameters
*Linksys is a wholly owned subsidiary of Cisco Systems, Inc.
Router Configuration : Example
Port Forwarding:
Router‟s SIP an RTP Port
forwarded to Private IP of
SETU ATA
*Linksys is a wholly owned subsidiary of Cisco Systems, Inc.
SIP Trunk Parameters : Source Port IP
Address
Program the Source Port IP Address as VoIP Ethernet Port
IP Address if WAN Port directly provided Public IP,
Incase of Behind Router Application program STUN
fetched or Router‟s Public IP as per configuration selected
SIP Extension General Parameters :
Source Port IP Address
Program the Source Port IP Address as VoIP Ethernet Port
IP Address if WAN Port directly provided Public IP,
Incase of Behind Router Application program STUN
fetched or Router‟s Public IP as per configuration selected
VoIP Port Parameters
Despite the advantages that SIP over TCP offers, it is more
common to use UDP to transport SIP messages. Enable this
flag if you want to receive SIP messages over TCP.
Transport of SIP messages
To be able to send SIP messages over TCP,
Configure „TCP‟ or „TCP(Fallback to UDP)‟.
VoIP Port Parameters
SIP UDP Port: This Port defines the port on which the VoIP Port
of ETERNITY listens for SIP messages transported over UDP
(Range:1024-65535; default Port:5060)
SIP TCP Port: Port on which the VoIP Port of ETERNITY listens
for SIP messages transported over TCP
(Range:1025-65535; default Port:5060)
RTP Listening Port: Port on which the VoIP Port of ETERNITY
listens for RTP packets (Range:1024-65278 : Default :8000)
UDP NAT Keep Alive
When VoIP port is connected behind a NAT router and
SIP messages are transported over UDP, UDP NAT keep
alive messages must be sent to refresh the UDP binding
in NAT Router
TCP NAT Keep Alive
When VoIP port is connected behind a NAT router and
SIP messages are transported over TCP, TCP NAT
keep alive messages must be sent to refresh the TCP
binding in NAT Router
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