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IP & VoIP fundamentals
VOIP Client & Server Capabilities
IP based Products overview
VoIP
Agenda
IP Basics
VoIP
SIP
SIP Trunks
SIP Extensions
VoIP Port Configuration
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 addressing Schemes
IP v4
(32 Bits)
IPv6
(128 Bits)
IP v4
(32 Bits)
IPv6
(128 Bits)
IP Ranges of Different Classes
11010001.11011100.11001001.0111001
Decimal : 209.156.201.113
4,294,467,295
IP Addresses
3.4 * 10^36
IP Addresses
E.g.:11010001.11011100.11001001.01110001.11010
001.11011100.110011001.01110001.11010001.1101
1100.11001001.01110001.11010001.11011100.1100
1001.01110001
Decimal :
A524:72D3:2C80:DD02:00029:EC7A:002B:EA73
IPv4 Ranges of Different Classes
0.0.0.0 to 127.255.255.255
Supports 16 million hosts on each of 128 networks
128.0.0.0 to 191.255.255.255
Supports 65,536 hosts on each of 16,384 networks
192.0.0.0 to 223.255.255.255
Supports 256 hosts on each of 2 million networks
224.0.0.0 to 239.255.255.255
Reserved for multicast groups
240.0.0.0 to 254.255.255.255
Reserved for future use, Research, Development Purposes
Class A
Class B
Class C
Class D
Class E
Private IPv4 Address Range
Class A
Class B
Class C
10.0.0.0 to 10.255.255.255
Subnet : 255.0.0.0
2^8 Networks & 2^24 Hosts
172.16.0.0 to 172.31.255.255
Subnet : 255.255.0.0
2^16 Networks & 2^16 Hosts
192.168.0.0 to 192.168.255.255
Subnet : 255.255.255.0
2^24 Networks & 2^8 Hosts
Agenda
IP Basics
VoIP
SIP
SIP Trunks
SIP Extensions
VoIP Port Configuration
Introduction to VoIP
Introduction to VoIP
What is VoIP?
Voice over Internet Protocol is a technique which is
used in delivery of voice communication sessions over
internet
In VoIP calling, voice is first converted into digital
signals or IP packets and then transferred over internet
VoIP Devices
IP Phones
ATA
The phone able
to connect itself
directly to the
internet for VoIP
communication
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
Soft IP Phone
Mobile Phone
PC Based soft IP PBX with
PCI based Hardware for
PSTN interfaces PC based Soft IP
phones
Mobile Phones with
VoIP client software
Agenda
IP Basics
VoIP
SIP
SIP Trunks
SIP Extensions
VoIP Port Configuration
Agenda
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
For
transportation,
SIP uses TCP UDP
OR
Protocol it
uses
SIP V2.0
SIP
RTP
SIP
Real-Time Transport Protocol
is used in SIP for real media
transfer (Voice, Video etc.)
SIP Messages
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.
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
Voice Coders
Codec & Bit Rate (Kbps) Bandwidth Ethernet (Kbps)
G.711 (64 Kbps)
G.729 (8 Kbps)
G.723.1 (6.3 Kbps)
G.723.1 (5.3 Kbps)
G.726 (32 Kbps)
G.726 (24 Kbps)
G.728 (16 Kbps)
87.2 Kbps
31.2 Kbps
21.9 Kbps
20.8 Kbps
55.2 Kbps
47.2 Kbps
31.5 Kbps
VoIP Channel
Number of VoIP channels indicated the total number of Simultaneous
VoIP calls
that can be made using a Particular SIP Device
VoIP 16 Card VoIP 32 Card
16 Channels 32 Channels
SETU ATA
Ranges
SPARSH VP248
2 VoIP Channels
VoIP Port Configuration
Agenda
IP Basics
VoIP
SIP
SIP Trunks
SIP Extensions
What is SIP Trunk?
VoIP calls can be Initiated after suitable programming of SIP Trunk number
in the
OG Trunk Bundle Group
SIP TRUNK IP CLOUD
SIP Trunks V/S VoIP Channels
SIP Trunks VoIP Channels
A medium to carry
VoIP calls from a
SIP device
Simultaneous calls that can be
done for that device depends
on the VoIP channels provided
SIP Trunks- Client Application
Types of VoIP Calling/ SIP Trunks
Peer-to-Peer Calling Proxy calling
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.
SIP Device
INVITE sip:203.88.142.218
100 Trying - 180 Ringing - 200 OK
ACK
RTP session (voice, video, etc)
203.88.143.218 204.88.142.218
Peer-to-Peer Calling
System A System B
Phone 102 Phone 402
Programming IP Details in ETERNITY
SIP Trunk : Peer to Peer in ETERNITY
SIP Trunk : Peer to Peer in ETERNITY
Public IP
INTERNET
115.118.161.163
Users can
directly access
the device over
internet
(Public IP Address)
P2P Call : Both Devices are in Public IP
203.88.143.75
Public IP
203.88.142.218
Public IP
INTERNET
P2P Call: One Device is on Public IP and
Other Device installed behind NAT
192.168.1.254
Internet
IP: 192.168.1.2
G/W : 192.168.1.254
Router separates
Private and Public
Network
Private IP
Public IP
203.88.142.218
Port Forward in
Router
LAN
WAN
203.88.142.220
SIP (102@203.88.142.219)
INVITE SDP (402@220.225.50.115)
100 Trying
180 Ringing
200 OK
ACK
Media Session (RTP)
BYE
200 OK for BYE
SIP (402@220.225.50.115)
Peer- to- Peer Call Flow
Proxy Calling
Making VoIP calls through proxy server is called proxy calling
Proxy Server: abc.com
Client 1
SIP ID 401
Client 2
SIP ID 402
Client 3
SIP ID 403
401 calling 402
SIP Device
Requirement for Proxy Calling
Proxy server authenticates the clients for outgoing calls through it
What is required
for
authentication?
SIP ID
Authentication ID
Authentication
Password
Registrar Server
Address
Registrar Server
port
SIP Trunk : Proxy
SIP Trunk : Proxy
Proxy Calling : Call Flow
SIP Agent (456@pqr.com) SIP Server (abc.com)
INVITE SDP (123@xyz.com)
INVITE SDP (123@xyz.com)
100 Trying
180 Ringing
180 Ringing
200 OK (456@pqr.com)
200 OK 456@pqr.com
ACK
ACK
407 Proxy Authentication
required ACK
Authenticated Media Session (RTP)
BYE
200 OK for BYE
100 Trying
VoIP Port Configuration
Agenda
IP Basics
VoIP
SIP
SIP Trunks
SIP Extensions
Range of MATRIX Products with VoIP Interface
SIP Extensions- Server capabilities
ETERNITY ME/GE/PE VoIP Server Card, ETERNITY NE VoIP Server Module & SAPEX IP
PBX Server have Server Capabilities
They behave as a Proxy Server and provides SIP
Accounts to Other SIP Devices
SIP extension user
of the IP PBX and can avail the System resource as well as make calls to other such
users
Configuring SIP Extensions
SIP extensions
can be
registered to
OR
local IP of Server incase it is in
the same network
Public Internet
N/W provided by the
Server
Configuring SIP Extensions
Server End Client End
SIP ID
Authentication ID
Authentication Password
SIP ID
Authentication ID
Authentication Password
Registrar Server Address
SIP Extension Settings : ETERNITY
VoIP Port Configuration
Agenda
IP Basics
VoIP
SIP
SIP Trunks
SIP Extensions
Range of MATRIX Products with VoIP Interface
VoIP Port Configuration in ETERNITY
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
LAN Port Configuration
Hardware Slot &
Port Offset
Customization
is Not Possible
Name Can Be
Assigned Just
For Easy
Identification
MAC Address
Of LAN Port
Configure IP Address
And Subnet Mask For
LAN Port
LAN Port Doesn’t Support
DHCP Connection
MAC Address Of
WAN Port
Customization
is Not Possible
Enable/Disable
MAC Cloning
Using This Flag
Configure Clone
MAC Address
WAN Port Configuration
WAN Port Configuration
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
WAN Port Configuration
When Connection Type : Static
Configure the 32 bit IP Address,
Subnet Mask & Gateway Address
as provided by the network
administrator or ISP
WAN Port Configuration
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
Dynamic DNS Configuration
Program the ‘User-ID’
‘Password’ 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
DDNS option will be useful only if the Internet
Connection Type is DHCP or PPPoE
Steps to configure the DynDNS
Open the
DynDNS
Server on
dyndns.org
Enter the Username & Password.
Create an account if you are
accessing it for the 1st time
*‘DYN DNS’ is trademark of Dynamic Network Services INC, USA
Steps to configure the DynDNS
Go to My Services -
Add Host Name
Steps to configure the DynDNS
Configure the
parameters for
a new Host
Name
Steps to configure the DynDNS
Select VoIP &
Add to Cart
Steps to configure the DynDNS
The dyndns host will be
created. Click on Next
Steps to configure the DynDNS
Total Hosts created
STUN
Simple Traversal of UDP through NATs
UDP (User Datagram Protocol) is a Network Protocol for
Transmission of Data
STUN
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
Illustration of STUN
STUN Request
STUN Request
STUN Response
To:115.118.161.163:5060
Payload:115.118.161.163:5060
STUN Response
To: 192.168.50.161:5060
Payload:115.118.161.163:5060
Source:192.168.50.161:5060
Source: 115.118.161.163:5060
STUN Server
IP
Device
With
inbuilt
STUN
client
Configuring STUN
Select this option
only if you have not
forwarded the SIP
Listening Port in the
Router. If flag is
“Enabled” then
System will use the
SIP listening Port
information
provided by the
STUN Server
Program the STUN Server
IP Address here
Program the STUN
Server port here
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
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
192.168.200.210
Internet
SETU ATA
IP: 192.168.200.195
G/W : 192.168.200.210
Router separates
Private and Public
Network
Private IP
Public IP
203.88.142.218
Port Forward in
Router
LAN port of Router WAN
203.88.142.221
Router Configuration : Example
Router’s
Network
Parameter
s
Router Configuration : Example
Port Forwarding:
Router’s SIP and
RTP Ports are
forwarded to
Private IP of SETU
ATA
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
Some Concerns related to
SIP Extension and VoIP Channels
When SIP Extension makes a call to another SIP Extension a total of Two VoIP
Channels will be consumed
When a SIP Extension makes a call Using any other Trunk except SIP Trunk of the
System only one VoIP channel will be consumed
When a SIP Extension makes a OG call using a SIP Trunk of the System Two VoIP
Channels will be consumed
When a normal DKP/SLT Extension of a system make OG call using SIP Trunk of the
system only one VoIP channel will be consumed
VoIP Port Configuration
Agenda
IP Basics
VoIP
SIP
SIP Trunks
SIP Extensions
Range of MATRIX Products with VoIP Interface
ETERNITY ME / GE / PE VoIP Interface:
VoIP Server Card
Hardware ME
10S
ME
16S
GE 6S GE 12S PE 3SP PE 6SP
Maximum VoIP
Channels /
VOIP calls per card
32 32 32 32 16 16
VoIP/ SIP Trunks 32 32 16 16 16 16
SIP Extensions 999 999 500 500 50 50
ETERNITY NE: Hybrid IP PBX with Server
Capabilities
8 VoIP Channels
8 SIP Trunks
Up to 16 IP Extensions
SAPEX : Pure IP-PBX Server
Up to 500 IP Extensions
10 SIP Trunks
MATRIX IP Phones
SPARSH VP248P/PE
2 VoIP Channels
3 SIP Trunks
SPARSH VP248S/SE
2 VoIP Channels
3 SIP Trunks
ATAs ( Analog Terminal Adaptors)
SETU ATA 2S
2 FXS
2 VoIP Channels
3 SIP Trunks
SETU ATA 1S
1 FXS
2 VoIP Channels
3 SIP Trunks
SETU ATA 211
1 FXO
1 FXS
2 VoIP Channels
3 SIP Trunks
SETU ATA 211G
1 FXS
1 GSM
2 VoIP Channels
3 SIP Trunks
MATRIX VoIP Gateways
Configuration
VoIP
Channels
FXO
Ports
FXS
Ports
SETU VFX404 4 0 4
SETU VFX440 4 4 0
SETU VFX808 8 0 8
SETU VFX880 8 8 0
9 SIP Trunks
MATRIX VoIP Gateways
Configuration
VoIP
Channels
FXO
Ports
FXS
Ports
SETU VFXTH0016 16 0 16
SETU VFXTH0032 32 0 32
SETU VFXTH1600 16 16 0
SETU VFXTH2400 24 24 0
SETU VFXTH3200 32 32 0
SETU VFXTH0808 16 08 08
SETU VFXTH1616 32 16 16
32 SIP Trunks
MATRIX VoIP Gateways
SETU VTEP321
VoIP to T1/E1/PRI Gateway
Up to 32 VoIP Channels
32 SIP Trunks
1 T1/E1 PRI Port
Network Clock Synchronization
SETU VGFX8422
SETU VGFX8404
SETU VGFX8440
VoIP-GSM-FXO-FXS
8 VoIP Channels
9 SIP Trunks
4-GSM Ports
2/4-FXO Ports
2/4-FXS Ports
MATRIX VoIP Gateways
SETU VGB842
VoIP- GSM-ISDN BRI Gateway
Plug-n-Play Configuration
8 VoIP Channels
4 GSM Channel
2 ISDN BRI Port
Network Clock Synchronization
SETU VBR42
VoIP to ISDN BRI Gateway
4 VoIP Channels
2 ISDN BRI Ports
2 Ethernet Ports
Network Clock Synchronization
IP and VoIP Fundamentals

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IP and VoIP Fundamentals

  • 2. IP & VoIP fundamentals VOIP Client & Server Capabilities IP based Products overview VoIP
  • 3. Agenda IP Basics VoIP SIP SIP Trunks SIP Extensions VoIP Port Configuration
  • 4. 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 addressing Schemes IP v4 (32 Bits) IPv6 (128 Bits)
  • 5. IP v4 (32 Bits) IPv6 (128 Bits) IP Ranges of Different Classes 11010001.11011100.11001001.0111001 Decimal : 209.156.201.113 4,294,467,295 IP Addresses 3.4 * 10^36 IP Addresses E.g.:11010001.11011100.11001001.01110001.11010 001.11011100.110011001.01110001.11010001.1101 1100.11001001.01110001.11010001.11011100.1100 1001.01110001 Decimal : A524:72D3:2C80:DD02:00029:EC7A:002B:EA73
  • 6. IPv4 Ranges of Different Classes 0.0.0.0 to 127.255.255.255 Supports 16 million hosts on each of 128 networks 128.0.0.0 to 191.255.255.255 Supports 65,536 hosts on each of 16,384 networks 192.0.0.0 to 223.255.255.255 Supports 256 hosts on each of 2 million networks 224.0.0.0 to 239.255.255.255 Reserved for multicast groups 240.0.0.0 to 254.255.255.255 Reserved for future use, Research, Development Purposes Class A Class B Class C Class D Class E
  • 7. Private IPv4 Address Range Class A Class B Class C 10.0.0.0 to 10.255.255.255 Subnet : 255.0.0.0 2^8 Networks & 2^24 Hosts 172.16.0.0 to 172.31.255.255 Subnet : 255.255.0.0 2^16 Networks & 2^16 Hosts 192.168.0.0 to 192.168.255.255 Subnet : 255.255.255.0 2^24 Networks & 2^8 Hosts
  • 8. Agenda IP Basics VoIP SIP SIP Trunks SIP Extensions VoIP Port Configuration
  • 11. What is VoIP? Voice over Internet Protocol is a technique which is used in delivery of voice communication sessions over internet In VoIP calling, voice is first converted into digital signals or IP packets and then transferred over internet
  • 12. VoIP Devices IP Phones ATA The phone able to connect itself directly to the internet for VoIP communication 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
  • 13. VoIP Devices Soft Switch Soft IP Phone Mobile Phone PC Based soft IP PBX with PCI based Hardware for PSTN interfaces PC based Soft IP phones Mobile Phones with VoIP client software
  • 14. Agenda IP Basics VoIP SIP SIP Trunks SIP Extensions VoIP Port Configuration Agenda
  • 16. 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 For transportation, SIP uses TCP UDP OR Protocol it uses SIP V2.0
  • 17. SIP RTP SIP Real-Time Transport Protocol is used in SIP for real media transfer (Voice, Video etc.)
  • 18. SIP Messages 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.
  • 19. 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
  • 20. Voice Coders Codec & Bit Rate (Kbps) Bandwidth Ethernet (Kbps) G.711 (64 Kbps) G.729 (8 Kbps) G.723.1 (6.3 Kbps) G.723.1 (5.3 Kbps) G.726 (32 Kbps) G.726 (24 Kbps) G.728 (16 Kbps) 87.2 Kbps 31.2 Kbps 21.9 Kbps 20.8 Kbps 55.2 Kbps 47.2 Kbps 31.5 Kbps
  • 21. VoIP Channel Number of VoIP channels indicated the total number of Simultaneous VoIP calls that can be made using a Particular SIP Device VoIP 16 Card VoIP 32 Card 16 Channels 32 Channels SETU ATA Ranges SPARSH VP248 2 VoIP Channels
  • 22. VoIP Port Configuration Agenda IP Basics VoIP SIP SIP Trunks SIP Extensions
  • 23. What is SIP Trunk? VoIP calls can be Initiated after suitable programming of SIP Trunk number in the OG Trunk Bundle Group SIP TRUNK IP CLOUD
  • 24. SIP Trunks V/S VoIP Channels SIP Trunks VoIP Channels A medium to carry VoIP calls from a SIP device Simultaneous calls that can be done for that device depends on the VoIP channels provided
  • 25. SIP Trunks- Client Application Types of VoIP Calling/ SIP Trunks Peer-to-Peer Calling Proxy calling
  • 26. 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. SIP Device
  • 27. INVITE sip:203.88.142.218 100 Trying - 180 Ringing - 200 OK ACK RTP session (voice, video, etc) 203.88.143.218 204.88.142.218 Peer-to-Peer Calling System A System B Phone 102 Phone 402
  • 28. Programming IP Details in ETERNITY
  • 29. SIP Trunk : Peer to Peer in ETERNITY
  • 30. SIP Trunk : Peer to Peer in ETERNITY
  • 31. Public IP INTERNET 115.118.161.163 Users can directly access the device over internet (Public IP Address)
  • 32. P2P Call : Both Devices are in Public IP 203.88.143.75 Public IP 203.88.142.218 Public IP INTERNET
  • 33. P2P Call: One Device is on Public IP and Other Device installed behind NAT 192.168.1.254 Internet IP: 192.168.1.2 G/W : 192.168.1.254 Router separates Private and Public Network Private IP Public IP 203.88.142.218 Port Forward in Router LAN WAN 203.88.142.220
  • 34. SIP (102@203.88.142.219) INVITE SDP (402@220.225.50.115) 100 Trying 180 Ringing 200 OK ACK Media Session (RTP) BYE 200 OK for BYE SIP (402@220.225.50.115) Peer- to- Peer Call Flow
  • 35. Proxy Calling Making VoIP calls through proxy server is called proxy calling Proxy Server: abc.com Client 1 SIP ID 401 Client 2 SIP ID 402 Client 3 SIP ID 403 401 calling 402 SIP Device
  • 36. Requirement for Proxy Calling Proxy server authenticates the clients for outgoing calls through it What is required for authentication? SIP ID Authentication ID Authentication Password Registrar Server Address Registrar Server port
  • 37. SIP Trunk : Proxy
  • 38. SIP Trunk : Proxy
  • 39. Proxy Calling : Call Flow SIP Agent (456@pqr.com) SIP Server (abc.com) INVITE SDP (123@xyz.com) INVITE SDP (123@xyz.com) 100 Trying 180 Ringing 180 Ringing 200 OK (456@pqr.com) 200 OK 456@pqr.com ACK ACK 407 Proxy Authentication required ACK Authenticated Media Session (RTP) BYE 200 OK for BYE 100 Trying
  • 40. VoIP Port Configuration Agenda IP Basics VoIP SIP SIP Trunks SIP Extensions Range of MATRIX Products with VoIP Interface
  • 41. SIP Extensions- Server capabilities ETERNITY ME/GE/PE VoIP Server Card, ETERNITY NE VoIP Server Module & SAPEX IP PBX Server have Server Capabilities They behave as a Proxy Server and provides SIP Accounts to Other SIP Devices SIP extension user of the IP PBX and can avail the System resource as well as make calls to other such users
  • 42. Configuring SIP Extensions SIP extensions can be registered to OR local IP of Server incase it is in the same network Public Internet N/W provided by the Server
  • 43. Configuring SIP Extensions Server End Client End SIP ID Authentication ID Authentication Password SIP ID Authentication ID Authentication Password Registrar Server Address
  • 45. VoIP Port Configuration Agenda IP Basics VoIP SIP SIP Trunks SIP Extensions Range of MATRIX Products with VoIP Interface
  • 46. VoIP Port Configuration in ETERNITY 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
  • 47. LAN Port Configuration Hardware Slot & Port Offset Customization is Not Possible Name Can Be Assigned Just For Easy Identification MAC Address Of LAN Port Configure IP Address And Subnet Mask For LAN Port LAN Port Doesn’t Support DHCP Connection
  • 48. MAC Address Of WAN Port Customization is Not Possible Enable/Disable MAC Cloning Using This Flag Configure Clone MAC Address WAN Port Configuration
  • 49. WAN Port Configuration 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
  • 50. WAN Port Configuration When Connection Type : Static Configure the 32 bit IP Address, Subnet Mask & Gateway Address as provided by the network administrator or ISP
  • 51. WAN Port Configuration 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
  • 52. Dynamic DNS Configuration Program the ‘User-ID’ ‘Password’ 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 DDNS option will be useful only if the Internet Connection Type is DHCP or PPPoE
  • 53. Steps to configure the DynDNS Open the DynDNS Server on dyndns.org Enter the Username & Password. Create an account if you are accessing it for the 1st time *‘DYN DNS’ is trademark of Dynamic Network Services INC, USA
  • 54. Steps to configure the DynDNS Go to My Services - Add Host Name
  • 55. Steps to configure the DynDNS Configure the parameters for a new Host Name
  • 56. Steps to configure the DynDNS Select VoIP & Add to Cart
  • 57. Steps to configure the DynDNS The dyndns host will be created. Click on Next
  • 58. Steps to configure the DynDNS Total Hosts created
  • 59. STUN Simple Traversal of UDP through NATs UDP (User Datagram Protocol) is a Network Protocol for Transmission of Data
  • 60. STUN 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
  • 61. Illustration of STUN STUN Request STUN Request STUN Response To:115.118.161.163:5060 Payload:115.118.161.163:5060 STUN Response To: 192.168.50.161:5060 Payload:115.118.161.163:5060 Source:192.168.50.161:5060 Source: 115.118.161.163:5060 STUN Server IP Device With inbuilt STUN client
  • 62. Configuring STUN Select this option only if you have not forwarded the SIP Listening Port in the Router. If flag is “Enabled” then System will use the SIP listening Port information provided by the STUN Server Program the STUN Server IP Address here Program the STUN Server port here
  • 63. 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
  • 64. Router’s Public IP Address Program the Static Router’s Public IP Address here
  • 65. P2P Call One Device is on Public IP and Other Device installed behind NAT 192.168.200.210 Internet SETU ATA IP: 192.168.200.195 G/W : 192.168.200.210 Router separates Private and Public Network Private IP Public IP 203.88.142.218 Port Forward in Router LAN port of Router WAN 203.88.142.221
  • 66. Router Configuration : Example Router’s Network Parameter s
  • 67. Router Configuration : Example Port Forwarding: Router’s SIP and RTP Ports are forwarded to Private IP of SETU ATA
  • 68. 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
  • 69. Some Concerns related to SIP Extension and VoIP Channels When SIP Extension makes a call to another SIP Extension a total of Two VoIP Channels will be consumed When a SIP Extension makes a call Using any other Trunk except SIP Trunk of the System only one VoIP channel will be consumed When a SIP Extension makes a OG call using a SIP Trunk of the System Two VoIP Channels will be consumed When a normal DKP/SLT Extension of a system make OG call using SIP Trunk of the system only one VoIP channel will be consumed
  • 70. VoIP Port Configuration Agenda IP Basics VoIP SIP SIP Trunks SIP Extensions Range of MATRIX Products with VoIP Interface
  • 71. ETERNITY ME / GE / PE VoIP Interface: VoIP Server Card Hardware ME 10S ME 16S GE 6S GE 12S PE 3SP PE 6SP Maximum VoIP Channels / VOIP calls per card 32 32 32 32 16 16 VoIP/ SIP Trunks 32 32 16 16 16 16 SIP Extensions 999 999 500 500 50 50
  • 72. ETERNITY NE: Hybrid IP PBX with Server Capabilities 8 VoIP Channels 8 SIP Trunks Up to 16 IP Extensions
  • 73. SAPEX : Pure IP-PBX Server Up to 500 IP Extensions 10 SIP Trunks
  • 74. MATRIX IP Phones SPARSH VP248P/PE 2 VoIP Channels 3 SIP Trunks SPARSH VP248S/SE 2 VoIP Channels 3 SIP Trunks
  • 75. ATAs ( Analog Terminal Adaptors) SETU ATA 2S 2 FXS 2 VoIP Channels 3 SIP Trunks SETU ATA 1S 1 FXS 2 VoIP Channels 3 SIP Trunks SETU ATA 211 1 FXO 1 FXS 2 VoIP Channels 3 SIP Trunks SETU ATA 211G 1 FXS 1 GSM 2 VoIP Channels 3 SIP Trunks
  • 76. MATRIX VoIP Gateways Configuration VoIP Channels FXO Ports FXS Ports SETU VFX404 4 0 4 SETU VFX440 4 4 0 SETU VFX808 8 0 8 SETU VFX880 8 8 0 9 SIP Trunks
  • 77. MATRIX VoIP Gateways Configuration VoIP Channels FXO Ports FXS Ports SETU VFXTH0016 16 0 16 SETU VFXTH0032 32 0 32 SETU VFXTH1600 16 16 0 SETU VFXTH2400 24 24 0 SETU VFXTH3200 32 32 0 SETU VFXTH0808 16 08 08 SETU VFXTH1616 32 16 16 32 SIP Trunks
  • 78. MATRIX VoIP Gateways SETU VTEP321 VoIP to T1/E1/PRI Gateway Up to 32 VoIP Channels 32 SIP Trunks 1 T1/E1 PRI Port Network Clock Synchronization SETU VGFX8422 SETU VGFX8404 SETU VGFX8440 VoIP-GSM-FXO-FXS 8 VoIP Channels 9 SIP Trunks 4-GSM Ports 2/4-FXO Ports 2/4-FXS Ports
  • 79. MATRIX VoIP Gateways SETU VGB842 VoIP- GSM-ISDN BRI Gateway Plug-n-Play Configuration 8 VoIP Channels 4 GSM Channel 2 ISDN BRI Port Network Clock Synchronization SETU VBR42 VoIP to ISDN BRI Gateway 4 VoIP Channels 2 ISDN BRI Ports 2 Ethernet Ports Network Clock Synchronization