This document provides an overview of IP and VoIP fundamentals including:
- IP basics such as IP addressing schemes (IPv4 and IPv6), IP ranges, and private IP address ranges.
- VoIP concepts such as devices, codecs, and channels.
- SIP including messages, responses, and call flows for peer-to-peer and proxy calling.
- SIP trunks and their differences from VoIP channels.
- SIP extensions and configuring them on a server.
- VoIP port configuration including LAN, WAN, DNS, and STUN/port forwarding.
- Matrix products that support VoIP including their VoIP channel and trunk capabilities.
I have described VoLTE IMS Architecture in simplified way . Are you also finding 3GPP Specs complicated & Complex for VoLTE IMS . It covers Role played by individual Networks Elements as mentioned below :-
# VoLTE SIP Handset : SIP Support , UAC , UAS , User Agent , SIP-UA
# Underlying LTE Network : MME , SGW , PGW , PCRF , HSS , Dedicated Bearer , QCI , Default Bearer
# IMS Core : SIP Servers , P-CSCF , I-CSCF , S-CSCF , TAS , MMTEL , BGw , MRF , ATCF , ATGW , IBCF , MGCF , IM-MGW , TrGW
# Voice Core or PSTN Network for Break-in or Break-out Calls
ims registration call flow procedure volte sipVikas Shokeen
This PDF , VoLTE IMS Registration tutorial covers IMS Registration sip procedure in depth & Provides extract of 3GPP / GSMA Specs , I am covering below call flow in Depth :-
- LTE Attach & Default Internet EPS bearer
- Role of QCI-1 ( Voice ) , QCI-5 (SIP Signaling) , QCI-6 to 9 (Internet)
- Default Vs Dedicated Bearer in LTE
- Default IMS EPS bearer in LTE
- SIP and IMS Registration
- TAS Registration
I have described VoLTE IMS Architecture in simplified way . Are you also finding 3GPP Specs complicated & Complex for VoLTE IMS . It covers Role played by individual Networks Elements as mentioned below :-
# VoLTE SIP Handset : SIP Support , UAC , UAS , User Agent , SIP-UA
# Underlying LTE Network : MME , SGW , PGW , PCRF , HSS , Dedicated Bearer , QCI , Default Bearer
# IMS Core : SIP Servers , P-CSCF , I-CSCF , S-CSCF , TAS , MMTEL , BGw , MRF , ATCF , ATGW , IBCF , MGCF , IM-MGW , TrGW
# Voice Core or PSTN Network for Break-in or Break-out Calls
ims registration call flow procedure volte sipVikas Shokeen
This PDF , VoLTE IMS Registration tutorial covers IMS Registration sip procedure in depth & Provides extract of 3GPP / GSMA Specs , I am covering below call flow in Depth :-
- LTE Attach & Default Internet EPS bearer
- Role of QCI-1 ( Voice ) , QCI-5 (SIP Signaling) , QCI-6 to 9 (Internet)
- Default Vs Dedicated Bearer in LTE
- Default IMS EPS bearer in LTE
- SIP and IMS Registration
- TAS Registration
Voice over Internet Protocol (VoIP) is a general term for a family of transmission technologies for delivery of voice communications over IP networks such as the Internet or other packet-switched networks.
VoLTE Flows and legacy CS network. Basic call routing to and from CS network using BGCF, MGCF, MGW. ENUM role in routing. IMS Cetralized Services (IMC) and SRVCC scenarios.
Marek Isalski, Faelix.net Ltd, describes the MikroTik range of routers and their applications, gives a pros and cons summary, and recommendations for budget provider edge deployment.
This presentation was presented at MUM Indonesia at Bali in 2008. Discussed about how to put extra layer of security into your MikroTik Router using Port Knocking mechanism.
Third revision of IMS signaling course. The lecture was part of the communication protocols class 2014 delivered to students from FIIT STU Bratislava, Slovakia and University Zilina, Slovakia.
VoLTE Voice over LTE Explained - Complete End to End VoLTE Overview - What is...Vikas Shokeen
Complete End to End Tutorial on Fundamentals & Basics of VoLTE , IMS Technology & VoLTE Overview ( Voice Over LTE )
- What is VoLTE
- Network Evolution to VoLTE
- How to Enable VoLTE in handset
- Differences between VoLTE & CSFB Call
- Voice call in LTE & VoLTE Networks
- Evolution of Voice Call
- VoLTE - Benefits for Users
- VoLTE - Benefits for Operators
- VoLTE Challenges
- Congestion handling for VoLTE Traffic
Voice over Internet Protocol (VoIP) is a general term for a family of transmission technologies for delivery of voice communications over IP networks such as the Internet or other packet-switched networks.
VoLTE Flows and legacy CS network. Basic call routing to and from CS network using BGCF, MGCF, MGW. ENUM role in routing. IMS Cetralized Services (IMC) and SRVCC scenarios.
Marek Isalski, Faelix.net Ltd, describes the MikroTik range of routers and their applications, gives a pros and cons summary, and recommendations for budget provider edge deployment.
This presentation was presented at MUM Indonesia at Bali in 2008. Discussed about how to put extra layer of security into your MikroTik Router using Port Knocking mechanism.
Third revision of IMS signaling course. The lecture was part of the communication protocols class 2014 delivered to students from FIIT STU Bratislava, Slovakia and University Zilina, Slovakia.
VoLTE Voice over LTE Explained - Complete End to End VoLTE Overview - What is...Vikas Shokeen
Complete End to End Tutorial on Fundamentals & Basics of VoLTE , IMS Technology & VoLTE Overview ( Voice Over LTE )
- What is VoLTE
- Network Evolution to VoLTE
- How to Enable VoLTE in handset
- Differences between VoLTE & CSFB Call
- Voice call in LTE & VoLTE Networks
- Evolution of Voice Call
- VoLTE - Benefits for Users
- VoLTE - Benefits for Operators
- VoLTE Challenges
- Congestion handling for VoLTE Traffic
Positive Hack Days. Gritsai. VOIP insecurities workshopPositive Hack Days
Участник получит представление об основе IP-телефонии, а также базовые навыки поиска уязвимостей на примере распространенных IP-PBX и абонентских устройств. Рассматриваются как типовые сетевые уязвимости, так и сложные случаи, обнаруживаемые в ходе анализа защищенности реальных сетей.
Matrix feature-rich ATAs offer connectivity to VoIP, GSM and POTS networks. An ATA user can plug standard analog telephone devices to the ATA and the analog device(s) will connect transparently to the IP and GSM networks. An ATA thus provides a user with the ease of using a standard telephone instrument, yet make VoIP and GSM calls. The ATAs can also be interfaced to existing PBX system, offering GSM and IP line to be shared among the PBX users.
SVR402: DirectAccess Technical Drilldown, Part 2 of 2: Putting it all together.Louis Göhl
Take a sprinkling of Windows 7, add Windows Server 2008 R2, IPv6 and IPsec and you have a solution that will allow direct access to your corporate network without the need for VPNs. Come to these demo-rich sessions and learn how to integrate DirectAccess into your environment. In Part 1 learn about IPv6 addressing, host configuration and transitioning technologies including 6to4, ISATAP, Teredo and IPHTTPS. Through a series of demos learn how to build an IPv6 Network and interoperate with IPv4 networks and hosts. In Part 2 we add the details of IPSec, and components that are only available with Windows 7 and Windows Server 2008 R2 to build the DirectAccess infrastructure. Learn how to control access to corporate resources and manage Internet connected PCs through group policy. Part 1 is highly recommended as a prerequisite for Part 2.
Matrix ETERNITY is a family of IP-PBXs with Universal Connectivity and Seamless Mobility. The ETERNITY IP-PBX offers built-in gateway capability to connect nearly all telecom interfaces like FXS, FXO, ISDN BRI, ISDN PRI, T1/E1, GSM and 3G.
SPARSH VP248 is a high-definition VoIP phone built with superior acoustics and elegant design to provide unsurpassed audio quality and rich user experience.
Based on open-standard SIP protocol, SPARSH VP248 is interoperable with any standard SIP infrastructure such as IP-PBX, SIP Proxies, Softswitches and Stand-alone applications.
SPARSH VP248 is designed for power users, knowledge workers and managers for quick access totheadvance system features and functions. A feature-packed IP phoneenables user to work efficiently with advance call handling capabilities
CÔNG TY CỔ PHẦN THẾ GIỚI TỔNG ĐÀI - NHÀ PHÂN PHỐI TỔNG ĐÀI CHUYÊN NGHIỆP
- Lắp đặt hệ thống CCTV chuyên nghiệp
- Lắp đặt hệ thống Camera giám sát.
- Triển khai hệ thống giám sát cho nhà xưởng.
- Cung cấp các giải pháp viễn thông cho doanh nghiệp
Vui lòng liên hệ:
Mr.Khoa : 0968.878.981
Email: khoa.nguyen@thegioitongdai.com.vn
Website: www.thegioitongdai.com.vn
Matrix Telecom Solutions: SETU VTEP - Fixed VoIP to T1/E1 PRI GatewayMatrix Comsec
Matrix SETU VTEP is a compact and dedicated gateway for VoIP to T1/E1 PRI network offering high-value communication experience to businesses of all size, Service Providers, Call Centers and simple but cost-effective solution for multi-location branch office communication. This intelligently designed gateway incorporates advanced features with multiple connectivity options to connect with a legacy communication system using T1/E1 or PRI signaling. SETU VTEP offers reliable and cost-effective solutions to the changing requirements of the business communication and offer customer value for money.
Connector Corner: Automate dynamic content and events by pushing a buttonDianaGray10
Here is something new! In our next Connector Corner webinar, we will demonstrate how you can use a single workflow to:
Create a campaign using Mailchimp with merge tags/fields
Send an interactive Slack channel message (using buttons)
Have the message received by managers and peers along with a test email for review
But there’s more:
In a second workflow supporting the same use case, you’ll see:
Your campaign sent to target colleagues for approval
If the “Approve” button is clicked, a Jira/Zendesk ticket is created for the marketing design team
But—if the “Reject” button is pushed, colleagues will be alerted via Slack message
Join us to learn more about this new, human-in-the-loop capability, brought to you by Integration Service connectors.
And...
Speakers:
Akshay Agnihotri, Product Manager
Charlie Greenberg, Host
GDG Cloud Southlake #33: Boule & Rebala: Effective AppSec in SDLC using Deplo...James Anderson
Effective Application Security in Software Delivery lifecycle using Deployment Firewall and DBOM
The modern software delivery process (or the CI/CD process) includes many tools, distributed teams, open-source code, and cloud platforms. Constant focus on speed to release software to market, along with the traditional slow and manual security checks has caused gaps in continuous security as an important piece in the software supply chain. Today organizations feel more susceptible to external and internal cyber threats due to the vast attack surface in their applications supply chain and the lack of end-to-end governance and risk management.
The software team must secure its software delivery process to avoid vulnerability and security breaches. This needs to be achieved with existing tool chains and without extensive rework of the delivery processes. This talk will present strategies and techniques for providing visibility into the true risk of the existing vulnerabilities, preventing the introduction of security issues in the software, resolving vulnerabilities in production environments quickly, and capturing the deployment bill of materials (DBOM).
Speakers:
Bob Boule
Robert Boule is a technology enthusiast with PASSION for technology and making things work along with a knack for helping others understand how things work. He comes with around 20 years of solution engineering experience in application security, software continuous delivery, and SaaS platforms. He is known for his dynamic presentations in CI/CD and application security integrated in software delivery lifecycle.
Gopinath Rebala
Gopinath Rebala is the CTO of OpsMx, where he has overall responsibility for the machine learning and data processing architectures for Secure Software Delivery. Gopi also has a strong connection with our customers, leading design and architecture for strategic implementations. Gopi is a frequent speaker and well-known leader in continuous delivery and integrating security into software delivery.
Smart TV Buyer Insights Survey 2024 by 91mobiles.pdf91mobiles
91mobiles recently conducted a Smart TV Buyer Insights Survey in which we asked over 3,000 respondents about the TV they own, aspects they look at on a new TV, and their TV buying preferences.
Software Delivery At the Speed of AI: Inflectra Invests In AI-Powered QualityInflectra
In this insightful webinar, Inflectra explores how artificial intelligence (AI) is transforming software development and testing. Discover how AI-powered tools are revolutionizing every stage of the software development lifecycle (SDLC), from design and prototyping to testing, deployment, and monitoring.
Learn about:
• The Future of Testing: How AI is shifting testing towards verification, analysis, and higher-level skills, while reducing repetitive tasks.
• Test Automation: How AI-powered test case generation, optimization, and self-healing tests are making testing more efficient and effective.
• Visual Testing: Explore the emerging capabilities of AI in visual testing and how it's set to revolutionize UI verification.
• Inflectra's AI Solutions: See demonstrations of Inflectra's cutting-edge AI tools like the ChatGPT plugin and Azure Open AI platform, designed to streamline your testing process.
Whether you're a developer, tester, or QA professional, this webinar will give you valuable insights into how AI is shaping the future of software delivery.
State of ICS and IoT Cyber Threat Landscape Report 2024 previewPrayukth K V
The IoT and OT threat landscape report has been prepared by the Threat Research Team at Sectrio using data from Sectrio, cyber threat intelligence farming facilities spread across over 85 cities around the world. In addition, Sectrio also runs AI-based advanced threat and payload engagement facilities that serve as sinks to attract and engage sophisticated threat actors, and newer malware including new variants and latent threats that are at an earlier stage of development.
The latest edition of the OT/ICS and IoT security Threat Landscape Report 2024 also covers:
State of global ICS asset and network exposure
Sectoral targets and attacks as well as the cost of ransom
Global APT activity, AI usage, actor and tactic profiles, and implications
Rise in volumes of AI-powered cyberattacks
Major cyber events in 2024
Malware and malicious payload trends
Cyberattack types and targets
Vulnerability exploit attempts on CVEs
Attacks on counties – USA
Expansion of bot farms – how, where, and why
In-depth analysis of the cyber threat landscape across North America, South America, Europe, APAC, and the Middle East
Why are attacks on smart factories rising?
Cyber risk predictions
Axis of attacks – Europe
Systemic attacks in the Middle East
Download the full report from here:
https://sectrio.com/resources/ot-threat-landscape-reports/sectrio-releases-ot-ics-and-iot-security-threat-landscape-report-2024/
Encryption in Microsoft 365 - ExpertsLive Netherlands 2024Albert Hoitingh
In this session I delve into the encryption technology used in Microsoft 365 and Microsoft Purview. Including the concepts of Customer Key and Double Key Encryption.
JMeter webinar - integration with InfluxDB and GrafanaRTTS
Watch this recorded webinar about real-time monitoring of application performance. See how to integrate Apache JMeter, the open-source leader in performance testing, with InfluxDB, the open-source time-series database, and Grafana, the open-source analytics and visualization application.
In this webinar, we will review the benefits of leveraging InfluxDB and Grafana when executing load tests and demonstrate how these tools are used to visualize performance metrics.
Length: 30 minutes
Session Overview
-------------------------------------------
During this webinar, we will cover the following topics while demonstrating the integrations of JMeter, InfluxDB and Grafana:
- What out-of-the-box solutions are available for real-time monitoring JMeter tests?
- What are the benefits of integrating InfluxDB and Grafana into the load testing stack?
- Which features are provided by Grafana?
- Demonstration of InfluxDB and Grafana using a practice web application
To view the webinar recording, go to:
https://www.rttsweb.com/jmeter-integration-webinar
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
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
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
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
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
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
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
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
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
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
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
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
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