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Hearty Welcome!
| Date post: | 20-Oct-2014 |
| Category: |
Technology |
| View: | 272 times |
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Hearty Welcome!
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.:11010001.11011100.11001001.01110001.11010001.11011100.110011001.01110001.11010001.11011100.11001001.01110001.11010001.11011100.11001001.01110001
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 ([email protected])
INVITE SDP ([email protected])
100 Trying
180 Ringing
200 OK
ACK
Media Session (RTP)
BYE
200 OK for BYE
SIP ([email protected])
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 ([email protected]) SIP Server (abc.com)
INVITE SDP ([email protected])
INVITE SDP ([email protected]) 100 Trying
180 Ringing 180 Ringing
200 OK ([email protected]) 200 OK [email protected]
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
Parameters
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
