Session Initiation Protocol (SIP) is an application layer protocol for setting up and managing multimedia communication sessions over IP networks. It allows users to initiate, modify and terminate multimedia sessions that include voice, video and messaging applications. SIP supports mobility through proxy servers that can forward calls to a user's current location. Common security threats to SIP include registration hijacking, message modification and denial of service attacks. Recommended security mechanisms include TLS for hop-by-hop security, S/MIME for end-to-end encryption, and digest authentication.
SIP - More than meets the eye
Speakers:
Ofer Cohen - VOIP Group Leader, LivePerson
Yossi Maimon - VOIP Technical Leader, LivePerson
An Introduction to the SIP protocol.
SIP Position in telecommunication networks and the content services.
What is SIP:
The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams. SIP can be used for two-party (unicast) or multiparty (multicast) sessions. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.
(Source: Wikipedia)
The presentation is a compiled assembly from the SIP RFC' s, and original works of Alan Johnston and Henry Sinnreich . It contains Sip Detailed , Call flows , Architecture descriptions , SIP services , sip security , sip programming.
The Session Initiation Protocol (SIP) is the dominant signaling protocol used in VoIP today. It is
responsible for the establishment, control and termination of sessions by exchanging ASCII-text-based
messages between the endpoints. This post goes through the basic components of SIP: messages and
logical entities.
To fully appreciate SIP you need to understand its routing capabilities and how they enable SIP to traverse a network. These capabilities also help SIP deal with common network issues such as NAT and firewalls. SIP's flexible routing also enables features like application composition, a very valuable asset when designing, implementing, and building a loosely coupled system.
This presentation is for those that are looking to get a deeper understanding of SIP. Perhaps you have been tasked to spin up a completely new SIP infrastructure at work? Then you really need to understand how SIP finds its way through a network. By understanding the routing decisions SIP makes, you will be successful in your next SIP endeavor.
Questions that will be answered:
- How does a SIP request traverse the network?
- How do we know which transport to use?
- How do responses find their way back?
- Any difference for in-dialog requests?
SIP - More than meets the eye
Speakers:
Ofer Cohen - VOIP Group Leader, LivePerson
Yossi Maimon - VOIP Technical Leader, LivePerson
An Introduction to the SIP protocol.
SIP Position in telecommunication networks and the content services.
What is SIP:
The Session Initiation Protocol (SIP) is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
The protocol defines the messages that are sent between peers which govern establishment, termination and other essential elements of a call. SIP can be used for creating, modifying and terminating sessions consisting of one or several media streams. SIP can be used for two-party (unicast) or multiparty (multicast) sessions. Other SIP applications include video conferencing, streaming multimedia distribution, instant messaging, presence information, file transfer, fax over IP and online games.
(Source: Wikipedia)
The presentation is a compiled assembly from the SIP RFC' s, and original works of Alan Johnston and Henry Sinnreich . It contains Sip Detailed , Call flows , Architecture descriptions , SIP services , sip security , sip programming.
The Session Initiation Protocol (SIP) is the dominant signaling protocol used in VoIP today. It is
responsible for the establishment, control and termination of sessions by exchanging ASCII-text-based
messages between the endpoints. This post goes through the basic components of SIP: messages and
logical entities.
To fully appreciate SIP you need to understand its routing capabilities and how they enable SIP to traverse a network. These capabilities also help SIP deal with common network issues such as NAT and firewalls. SIP's flexible routing also enables features like application composition, a very valuable asset when designing, implementing, and building a loosely coupled system.
This presentation is for those that are looking to get a deeper understanding of SIP. Perhaps you have been tasked to spin up a completely new SIP infrastructure at work? Then you really need to understand how SIP finds its way through a network. By understanding the routing decisions SIP makes, you will be successful in your next SIP endeavor.
Questions that will be answered:
- How does a SIP request traverse the network?
- How do we know which transport to use?
- How do responses find their way back?
- Any difference for in-dialog requests?
A short presentation on SIP Trunking. A background with SIP training and PSTN (T-1, ISDN PRI) and TCP/IP knowledge will be helpful. For more info connect with me at http://TrainingCity.com
Brief introduction into SIP protocol, how it works, common problems to solve. Tech. details about handshake, SIP Trunks and SIP trunking. Market research.
Firewalls, SIP Servers and SBC - What's the Differences?Alan Percy
There are a lot of building blocks that make up a reliable and secure voice network. Firewalls, SIP Servers and Session Border Controllers each play an important, but different, role in facilitating a secure and reliable network. Join us as we take a closer look at each in a modern voice network, answering the questions:
-What are they?
-Why do I need them?
-What do they do?
-What do they not do?
-Your questions
Each of the series will be recorded and library of recordings made available to registrants.
Who should view:
Network architects at service providers and enterprises, networking consultants, analysts and decision-makers involved in acquiring network technology
Download your FreeSBC software: www.freesbc.com
Frequently Asked Questions at: forums.freesbc.com
Other educational webinars at: freesbc.com/video-library
VoLTE Basic callflows in IMS network v2 - includes Registration, Basic VoLTE Call, SDP, Interconnect, Roaming, highlights important SIP headers for session routing and user identities.
Over the past 10 years the Session Initiation Protocol (SIP) has moved from the toy of researchers and academics to the de-facto standard for telephony and multimedia services in mobile and fixed networks.
Probably one of the most emotionally fraught discussions in the context of SIP was whether Session Border Controllers (SBC) are good or evil.
SIP was designed with the vision of revolutionizing the way communication services are developed, deployed and operated. Following the end-to-end spirit of the Internet SIP was supposed to turn down the walled gardens of PSTN networks and free communication services from the grip of large telecom operators. By moving the intelligence to the end systems, developers were supposed to be able to develop new communication services that will innovate the way we communicate with each other.
This was to be achieved without having to wait for the approval of the various telecommunication standardization groups such as ETSI or the support of incumbent telecoms.
Session border controllers are usually implemented as SIP Back-to-Back User Agents (B2BUA) that are placed between a SIP user agent and a SIP proxy. The SBC then acts as the contact point for both the user agents and the proxy. Thereby the SBC actually breaks the end-to-end behavior of SIP, which has led various people to deem the SBC as an evil incarnation of the old telecom way of thinking. Regardless of this opposition, SBCs have become a central part of any SIP deployment.
In this paper we will first give a brief overview of how SIP works and continue with a description of what SBCs do and the different use cases for deploying SBCs.
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
If you don't know what SIP is, what it is used for or why you should even care then this is the section for you. This presentation will go over the very basics of SIP and assumes no previous knowledge of SIP or really any other network experience either. Topics it will touch upon is:
- What SIP is all about
- SIP and sessions management
- Basic call flow
- Brief discussion of SIP messages
- SIP and audio
Since the dawn of the telephone operator switchboard, call routing has challenged network designers and managers with complex routing algorithms. While the technology has changed, the need remains the same.
Routing SIP traffic is one of the key functions of a Session Border Controller, directing traffic and working around network outages.
Continuing on the "How To" series of installation and provisioning sessions, this 30-minute educational webinar offers an opportunity to learn the fundamental and advanced SIP routing functions found in FreeSBC.
Topics covered in this session:
SIP Routing Architectures and Challenges
Basic SIP Routing Configuration
Dealing with Outages
Advanced SIP Routing Tables
Real-world Use Cases
Your Questions
Who would find most valuable:
Network architects at service providers and enterprises, networking consultants, analysts and decision-makers involved in acquiring network technology
Overview of VoIP (Voice over IP) and FoIP (Fax over IP) technologies like Session Initiation Protocol and H.323.
Even though voice over IP (VoIP) was hailed as a technological innovation, the idea to transport real-time traffic over TCP/IP networks was not new back in the 1990s when VoIP started being deployed in networks. Chapter 2.5 of the venerable RFC793 (TCP) shows both data oriented application traffic as well as voice being transported over IP based networks.
Nevertheless, VoIP puts high demands on signal and protocol processing capabilities so it became possible at reasonable costs only in the 1990s.
VoIP can be roughly split into two main functions. Signaling protocols like SIP (Session Initiation Protocol), H.323 and MGCP/H.248 are used to establish a conference session and the data path for transporting real-time voice data packets. SIP has largely supplanted H.323 in recent years to its simpler structure and packet sequences. MGCP and H.248 are mostly used in carrier backbone networks.
Protocols like RTP (Real Time Protocol) transport voice packets and provide the necessary information for receivers to equalize packet flow variations to provide a smooth playback of the original voice signal.
Voice codecs are one of the core functions of the data path. Voice compression reduces the bandwidth required to transport voice over an IP based network. Compression may be less of a concern in local area networks with gigabit speeds, on slower links like 3G (UMTS, LTE) it still makes a lot of sense.
The algorithms used in different codecs make use of various characteristics of the characteristics of human speech recognition. Redundant information is removed from the signals thus slightly reducing the quality, but greatly reducing the required bandwidth.
In VoIP networks, the echo problem is typically compounded by the increased delay incurred by packetization of voice signals. To counteract the echo problem, VoIP gear (hard phones, soft phones, gateways) include echo cancelers to remove echo signals from the transmit signal.
To transport facsimile over an IP based network, even more technology is needed. Facsimile protocols are very susceptible to delay and delay variation and thus need more compensation algorithms. Protocols like T.38 terminate facsimile protocols like T.30 (analog facsimile) and transport the fax images as digitized pictures over IP based networks.
A short presentation on SIP Trunking. A background with SIP training and PSTN (T-1, ISDN PRI) and TCP/IP knowledge will be helpful. For more info connect with me at http://TrainingCity.com
Brief introduction into SIP protocol, how it works, common problems to solve. Tech. details about handshake, SIP Trunks and SIP trunking. Market research.
Firewalls, SIP Servers and SBC - What's the Differences?Alan Percy
There are a lot of building blocks that make up a reliable and secure voice network. Firewalls, SIP Servers and Session Border Controllers each play an important, but different, role in facilitating a secure and reliable network. Join us as we take a closer look at each in a modern voice network, answering the questions:
-What are they?
-Why do I need them?
-What do they do?
-What do they not do?
-Your questions
Each of the series will be recorded and library of recordings made available to registrants.
Who should view:
Network architects at service providers and enterprises, networking consultants, analysts and decision-makers involved in acquiring network technology
Download your FreeSBC software: www.freesbc.com
Frequently Asked Questions at: forums.freesbc.com
Other educational webinars at: freesbc.com/video-library
VoLTE Basic callflows in IMS network v2 - includes Registration, Basic VoLTE Call, SDP, Interconnect, Roaming, highlights important SIP headers for session routing and user identities.
Over the past 10 years the Session Initiation Protocol (SIP) has moved from the toy of researchers and academics to the de-facto standard for telephony and multimedia services in mobile and fixed networks.
Probably one of the most emotionally fraught discussions in the context of SIP was whether Session Border Controllers (SBC) are good or evil.
SIP was designed with the vision of revolutionizing the way communication services are developed, deployed and operated. Following the end-to-end spirit of the Internet SIP was supposed to turn down the walled gardens of PSTN networks and free communication services from the grip of large telecom operators. By moving the intelligence to the end systems, developers were supposed to be able to develop new communication services that will innovate the way we communicate with each other.
This was to be achieved without having to wait for the approval of the various telecommunication standardization groups such as ETSI or the support of incumbent telecoms.
Session border controllers are usually implemented as SIP Back-to-Back User Agents (B2BUA) that are placed between a SIP user agent and a SIP proxy. The SBC then acts as the contact point for both the user agents and the proxy. Thereby the SBC actually breaks the end-to-end behavior of SIP, which has led various people to deem the SBC as an evil incarnation of the old telecom way of thinking. Regardless of this opposition, SBCs have become a central part of any SIP deployment.
In this paper we will first give a brief overview of how SIP works and continue with a description of what SBCs do and the different use cases for deploying SBCs.
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
If you don't know what SIP is, what it is used for or why you should even care then this is the section for you. This presentation will go over the very basics of SIP and assumes no previous knowledge of SIP or really any other network experience either. Topics it will touch upon is:
- What SIP is all about
- SIP and sessions management
- Basic call flow
- Brief discussion of SIP messages
- SIP and audio
Since the dawn of the telephone operator switchboard, call routing has challenged network designers and managers with complex routing algorithms. While the technology has changed, the need remains the same.
Routing SIP traffic is one of the key functions of a Session Border Controller, directing traffic and working around network outages.
Continuing on the "How To" series of installation and provisioning sessions, this 30-minute educational webinar offers an opportunity to learn the fundamental and advanced SIP routing functions found in FreeSBC.
Topics covered in this session:
SIP Routing Architectures and Challenges
Basic SIP Routing Configuration
Dealing with Outages
Advanced SIP Routing Tables
Real-world Use Cases
Your Questions
Who would find most valuable:
Network architects at service providers and enterprises, networking consultants, analysts and decision-makers involved in acquiring network technology
Overview of VoIP (Voice over IP) and FoIP (Fax over IP) technologies like Session Initiation Protocol and H.323.
Even though voice over IP (VoIP) was hailed as a technological innovation, the idea to transport real-time traffic over TCP/IP networks was not new back in the 1990s when VoIP started being deployed in networks. Chapter 2.5 of the venerable RFC793 (TCP) shows both data oriented application traffic as well as voice being transported over IP based networks.
Nevertheless, VoIP puts high demands on signal and protocol processing capabilities so it became possible at reasonable costs only in the 1990s.
VoIP can be roughly split into two main functions. Signaling protocols like SIP (Session Initiation Protocol), H.323 and MGCP/H.248 are used to establish a conference session and the data path for transporting real-time voice data packets. SIP has largely supplanted H.323 in recent years to its simpler structure and packet sequences. MGCP and H.248 are mostly used in carrier backbone networks.
Protocols like RTP (Real Time Protocol) transport voice packets and provide the necessary information for receivers to equalize packet flow variations to provide a smooth playback of the original voice signal.
Voice codecs are one of the core functions of the data path. Voice compression reduces the bandwidth required to transport voice over an IP based network. Compression may be less of a concern in local area networks with gigabit speeds, on slower links like 3G (UMTS, LTE) it still makes a lot of sense.
The algorithms used in different codecs make use of various characteristics of the characteristics of human speech recognition. Redundant information is removed from the signals thus slightly reducing the quality, but greatly reducing the required bandwidth.
In VoIP networks, the echo problem is typically compounded by the increased delay incurred by packetization of voice signals. To counteract the echo problem, VoIP gear (hard phones, soft phones, gateways) include echo cancelers to remove echo signals from the transmit signal.
To transport facsimile over an IP based network, even more technology is needed. Facsimile protocols are very susceptible to delay and delay variation and thus need more compensation algorithms. Protocols like T.38 terminate facsimile protocols like T.30 (analog facsimile) and transport the fax images as digitized pictures over IP based networks.
"Implementación práctica de TLS, SRTP y OpenVPN en Elastix para encriptar las comunicaciones"
Juan Almeida - Minga.ec, Noviembre 27 de 2013
Quito, Ecuador
This is a presentation designed for IP Telephony Site Partners and key Super Users. It identifies some of the key technologies and telephony chocies that need to be made in moving a University of Mlbourne staff member to the new IP Telephony system.
"Green" networks are a hot topic, and saving on power costs is a corporate mandate at many enterprises. Many aspects of an IP Telephony deployment offer unique challenges in this area. For example, many enterprises are unaware of the cost to power their wiring closets. These costs are likely to increase as enterprises deploy power over Ethernet into each wiring closets. Learn more about how to conserve power and reduce costs.
Enterprise communications systems have come a long way in the past 30 years. Now the important issues for developing an architecture for IP telephony deployment include: architectural & design attributes of each platform, customer benefits and advantages, the emerging role of unified communications, the evolution to a wireless mobile communications platform, current and developing communications standards and open source solutions. Understand the past--and the present--trends of communications systems.
SIP Trunking & Security in an Enterprise NetworkDan York
How secure are your VoIP systems as you deploy SIP-based systems in an enterprise environment? In this slide deck presented by VOIPSA Best Practices Chair Dan York at the Ingate SIP Trunking Seminars at ITEXPO September 17, 2008, Dan York walks through the security issues related to VoIP (with a focus on SIP trunking), the tools out there to attack/test VoIP systems, best practices and resources. (An audio recording of this session was made and will be available.)
With World IPv6 Launch happening June 6, 2012, production IPv6 network connectivity will be available to many more businesses and individuals. Major web sites and content providers will all enable IPv6 access to their content. Consumer electronics manufacturers are committing to providing IPv6-enabled devices.
What does this mean for SIP-based real-time communications? How well does SIP work with IPv6 today? What are the challenges to deployment and what steps can be taken to overcome those challenges? What should operators and vendors consider with regard to SIP and IPv6? What software, devices and tools are available to assist? And what case studies and other information is available?
In this session at SIPNOC 2012 on June 26, 2012, in Herndon, Virginia, Dan York discussed all of these points and provided concrete suggestions for moving forward with SIP and IPv6.
Can IPv6 and SIP really work together well? At the SIP Network Operators Conference (SIPNOC) on April 24, 2013, Dan York moderated a panel on this specific question and gave an update on IPv6 deployment and interaction with VoIP.
The Voice over Internet Protocol (VoIP). The VoIP is relatively new and is gaining more and more popularity as it offers a
wide range of features and is much more cost effective as compared to the traditional PSTN. But the VoIP brings with it certain
security threats which need to be resolved in order to make it a more reliable source of communication. Session Initiation Protocol
(SIP) today is considered the standard protocol for multimedia signaling, and the result is a very generic protocol. SIP is specified by
the IETF in RFC 3261. From a structural and functional perspective, SIP is application layer signaling text-based protocol used for
creating, modifying, and terminating multimedia communications sessions among Internet endpoints. Unfortunately, SIP-based
application services can suffer from various security threats as Denial of Service (DoS). attacks on a SIP based VoIP infrastructure that
can severely compromise its reliability. In contrast, little work is done to analyze the robustness and reliability of SIP severs under
DoS attacks. In this survey, we are discussing the DoS flooding attack on SIP server. Firstly, we present a brief overview about the SIP
protocol. Then, security attacks related to SIP protocol. After that, detection techniques of SIP flooding attack and various exploited
resources due to attack were discussed and finally the paper reviews previous work done on SIP based DoS attacks.
XMPP and SIP Presence Protocols for Messaging and Session Control.pptxGSCWU
XMPP is the Extensible Messaging and Presence Protocol, a set of open technologies for instant messaging, presence, multi-party chat, voice and video calls, collaboration, lightweight middleware, content syndication, and generalized routing of XML data.
XMPP was originally developed in the Jabber open-source community to provide an open, decentralized alternative to the closed instant messaging services at that time
The Session Initiation Protocol is a signaling protocol that enables the Voice Over Internet Protocol (VoIP) by defining the messages sent between endpoints and managing the actual elements of a call. SIP supports voice calls, video conferencing, instant messaging, and media distribution.
Rise of multimedia and network technologies, multimedia has become an indispensable feature on the Internet.
Animation, voice and video clips become more and more popular on the Internet. Multimedia networking products like Internet telephony, Internet TV, video conferencing have appeared on the market
CCIE Collaboration Bootcamp is designed to be a challenging five-day course for CCIE Collaboration candidates ready for CCIE Collaboration Lab Exam. This Bootcamp is designed for CCIE Collaboration candidates in the last months or weeks before their CCIE Collaboration Lab Exam. During the week students will tackle challenging full-day mock labs Monday through Thursday. Candidate will practice strategy, time management, learn test taking strategies and expose any weaknesses in order to resolve them before the lab exam. On the final day of the course
Kubernetes & AI - Beauty and the Beast !?! @KCD Istanbul 2024Tobias Schneck
As AI technology is pushing into IT I was wondering myself, as an “infrastructure container kubernetes guy”, how get this fancy AI technology get managed from an infrastructure operational view? Is it possible to apply our lovely cloud native principals as well? What benefit’s both technologies could bring to each other?
Let me take this questions and provide you a short journey through existing deployment models and use cases for AI software. On practical examples, we discuss what cloud/on-premise strategy we may need for applying it to our own infrastructure to get it to work from an enterprise perspective. I want to give an overview about infrastructure requirements and technologies, what could be beneficial or limiting your AI use cases in an enterprise environment. An interactive Demo will give you some insides, what approaches I got already working for real.
Neuro-symbolic is not enough, we need neuro-*semantic*Frank van Harmelen
Neuro-symbolic (NeSy) AI is on the rise. However, simply machine learning on just any symbolic structure is not sufficient to really harvest the gains of NeSy. These will only be gained when the symbolic structures have an actual semantics. I give an operational definition of semantics as “predictable inference”.
All of this illustrated with link prediction over knowledge graphs, but the argument is general.
The Art of the Pitch: WordPress Relationships and SalesLaura Byrne
Clients don’t know what they don’t know. What web solutions are right for them? How does WordPress come into the picture? How do you make sure you understand scope and timeline? What do you do if sometime changes?
All these questions and more will be explored as we talk about matching clients’ needs with what your agency offers without pulling teeth or pulling your hair out. Practical tips, and strategies for successful relationship building that leads to closing the deal.
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.
Essentials of Automations: Optimizing FME Workflows with ParametersSafe Software
Are you looking to streamline your workflows and boost your projects’ efficiency? Do you find yourself searching for ways to add flexibility and control over your FME workflows? If so, you’re in the right place.
Join us for an insightful dive into the world of FME parameters, a critical element in optimizing workflow efficiency. This webinar marks the beginning of our three-part “Essentials of Automation” series. This first webinar is designed to equip you with the knowledge and skills to utilize parameters effectively: enhancing the flexibility, maintainability, and user control of your FME projects.
Here’s what you’ll gain:
- Essentials of FME Parameters: Understand the pivotal role of parameters, including Reader/Writer, Transformer, User, and FME Flow categories. Discover how they are the key to unlocking automation and optimization within your workflows.
- Practical Applications in FME Form: Delve into key user parameter types including choice, connections, and file URLs. Allow users to control how a workflow runs, making your workflows more reusable. Learn to import values and deliver the best user experience for your workflows while enhancing accuracy.
- Optimization Strategies in FME Flow: Explore the creation and strategic deployment of parameters in FME Flow, including the use of deployment and geometry parameters, to maximize workflow efficiency.
- Pro Tips for Success: Gain insights on parameterizing connections and leveraging new features like Conditional Visibility for clarity and simplicity.
We’ll wrap up with a glimpse into future webinars, followed by a Q&A session to address your specific questions surrounding this topic.
Don’t miss this opportunity to elevate your FME expertise and drive your projects to new heights of efficiency.
UiPath Test Automation using UiPath Test Suite series, part 3DianaGray10
Welcome to UiPath Test Automation using UiPath Test Suite series part 3. In this session, we will cover desktop automation along with UI automation.
Topics covered:
UI automation Introduction,
UI automation Sample
Desktop automation flow
Pradeep Chinnala, Senior Consultant Automation Developer @WonderBotz and UiPath MVP
Deepak Rai, Automation Practice Lead, Boundaryless Group and UiPath MVP
Generating a custom Ruby SDK for your web service or Rails API using Smithyg2nightmarescribd
Have you ever wanted a Ruby client API to communicate with your web service? Smithy is a protocol-agnostic language for defining services and SDKs. Smithy Ruby is an implementation of Smithy that generates a Ruby SDK using a Smithy model. In this talk, we will explore Smithy and Smithy Ruby to learn how to generate custom feature-rich SDKs that can communicate with any web service, such as a Rails JSON API.
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
Accelerate your Kubernetes clusters with Varnish CachingThijs Feryn
A presentation about the usage and availability of Varnish on Kubernetes. This talk explores the capabilities of Varnish caching and shows how to use the Varnish Helm chart to deploy it to Kubernetes.
This presentation was delivered at K8SUG Singapore. See https://feryn.eu/presentations/accelerate-your-kubernetes-clusters-with-varnish-caching-k8sug-singapore-28-2024 for more details.
How world-class product teams are winning in the AI era by CEO and Founder, P...
SIP security in IP telephony
1.
2. INTRODUCTION
• Session Initiation Protocol (SIP) is a Requests For Comments
(RFC) of the Internet Engineering Task Force (IETF)
• First standardized in March 1999 in RFC 2543 (Obsolete)
• A second version in 2002 in RFC 3261
3. INTRODUCTION
• Today, the session initiation protocol (SIP) is the predominant
protocol for IP Telephony Signalling. This paper addresses IP
Telephony security issues - both current and future – focusing
on SIP.
• We summarize current activities regarding SIP
security, including recent developments in the research
community and standardization efforts within the IETF.
4. SIP OVERVIEW (1)
• ASCII based, signaling protocol
• Analogous to HTTP messages, SIP is a text base protocol.
• Works independent of the underlying network transmission
protocol and indifferent to media
5. SIP OVERVIEW (1)
It provides mechanisms to:
• Establish a session
• Maintain a session
• Modify and Terminate a session
• Session Initiation Protocol (SIP) is an application layer protocol, which is
used to establish, maintain and terminate multimedia session.
• These sessions may include voice, video, instant messaging.
6. SIP Components
System using SIP can be viewed in two Dimensions:
• Client/Server
• Individual Network Elements
7. SIP Components
Client : : A client is any network element that sends SIP
requests and receives SIP responses.
Server: A server is a network element that receives requests
in order to service them and sends back responses to those
requests.
• Example of Servers: Proxies, user agent servers, redirect
servers, and registrars.
8. SIP Components (2)
Two general categories of SIP are
User Agent (UA): Resides in every SIP end station
SIP Servers
9. SIP Components (2)
User Agent (UA)
Has two roles:
SIP User Agent Client(UAC): Issues SIP requests.
SIP User Agent Server (UAS): Receives SIP requests, and
Generates a response that accepts, rejects, or redirects the
request.
10. SIP Components (2)
SIP Servers
• Proxy Server: The proxy server is an intermediary entity that acts as both a server and a
client for the purpose of making requests on behalf of other clients. A proxy server primarily
plays the role of routing, meaning that its job is to ensure that a request is sent to another
entity closer to the targeted user.
• Redirect Server: Used during session initiation, Determine the address of the called
device, Returns this information to the calling device.
• Registrar Server: A registrar is a server that accepts REGISTER requests and places the
information it receives (the SIP address and associated IP address of the registering device) in
those requests into the location service for the domain it handles.
11. SIP Functions
Scalability
Functionality such as proxying, redirection, location, or registration can
reside in different physical servers.
Distributed functionality allows new processes to be added without
affecting other components.
Interoperability
An open standard
Can implement to communicate with other SIP based products
12. SIP Functions (2)
Mobility
• Supports user mobility by proxying and redirecting requests to a
user’s current location.
• The user can be using a PC at work, PC at home, wireless phone, IP
phone, or regular phone.
• Users must register their current location.
• Proxy servers will forward calls to the user’s current location.
• Example mobility applications include presence and call forking.
14. SIP CAPABILITIES
• Determine location of target points – Support address resolution, name
mapping, call redirection
• Determine media capabilities – SIP uses Session Description Protocol (SDP)
for this
• Determine availability – returns a message why the remote party cannot
be contacted
• Establish a session between end points – also support mid call
changes, changes of media characteristics or codec
• Handles termination of calls – transfer of calls
• Permits interaction between devices via signalling messages
15. SIP CAPABILITIES
• INVITE: Invite a user to join a call
• ACK: Confirm that a client has received a final response to an invite
• BYE: Terminates the call between two of the users on a call
• OPTIONS: Request information on the capabilities of a Server
• CANCEL: Ends a pending Request , but doesn’t end the call
• REGISTER: Provide the map of address resolution that lets the server know the location of the users.
16. Status Codes
1xxInformational
• 100 Trying
• 180 Ringing (ringing tone
played locally)
• 181 Call is Being
Forwarded
• 182 Queued
• 183 Session progress
2xxSuccess
• 200 ok
3xx Redirection
• 300 Multiple Choices
• 301 Moved Permanently
• 302 Moved Temporarily
• 380 Alternative server
4xxClient error
• 400 Bad Request
• 401 Unauthorized
• 403 Forbidden
• 404 Not Found
• 405 Bad Method
• 415 Unsupported
Content
• 420 Bad Extensions
• 482 Detected
• 486 Busy Here
5xxServer failure
• 500 Server Internal
Error
• 501 Not
Implemented
• 503 Unavailable
• 504 Timeout
6xxGlobal Failure
• 600 Busy Everywhere
• 603 Decline
• 604 Doesn’t Exist
• 606 Not Acceptable
18. SIP Headers
• Session Initiation Protocol (RFC3261) for call signaling
• Header format is similar to HTTPS
• UDP Port 5060 used (recommended)
• TCP is also allowed (required for SIPS)
• Responsible for connection setup and release:
INVITE, OK, ACK, BYE, CANCEL
• Registration service for mobile user agents: REGISTER
• Uses DNS for routing (RFC3263;)
19. SIP Headers
• Session Description Protocol (RFC 2327) for parameter exchange
• Body of SIP-Messages
• Looks (a little bit) like sendmail mail queue format
• Contact address (ip address, port #) c=IN IP4 172.16.1.127
• Codec m=audio 7078 RTP/AVP 8 0 2 102 100 97 101
• (Master)Key for SRTP k=clear:geheim
21. Breakdown of Header
INVITE :
message type
Address of called party
SIP version used by caller
Semicolon indicates start of URI parameters
Eg:- user=phone indicates call is for a phone number and not a SIP IP address
INVITE sip:09611000038@202.4.97.11 SIP/2.0
Via:
History of message’s path through network(s)
Helps to prevent looping and ensures replies route back to originator
Indicates the used transport protocol, ip address and port of sender
Via: SIP/2.0/UDP 172.16.1.127:6256;branch=z9hG4bK-d8754z-64630900441c9d08-1---
d8754z-;rport
22. SDP Headers
• Describes components of communication channel under negotiation
• Includes information about :
– Codecs
– Ports
– Streaming protocols
• Usually sent with INVITE and 200 OK in SIP based devices
• Describes how data stream is going to be support via Real Time Transport
Protocol (RTP, RFC 1889)
24. Security Attacks
Signaling Layer Attacks
• SIP Registration Hijacking: Attacker impersonates a valid UA to a
registrar himself as a valid user agent. so attacker can receive calls
for a valid user.
• Impersonating a Server: When an attacker impersonates a remote
server and user agent request are served by the attacker machine.
25. Security Attacks
Signaling Layer Attacks
• SIP Message Modification: If an attacker launches a man in the
middle attack and modify a message. Then attacker could lead the
caller to connect to malicious system.
• SIP Cancel / SIP BYE attack
• SIP DOS attack: In SIP attacker creates a bogus request that
contained a fake IP address and Via field in the SIP header contains
the identity of the target host.
26. Security Solutions
Two types of security solutions
End-to End security:
• In SIP end points can ensure end-to-end security to those messages which
proxy does not read, like SDP messages could be protected using S/MIME.
• Media is transferred directly, so end-to-end security is achieved by SRTP.
Hop-by-hop security
• TLS, IPSec.
27. SIP Security Mechanisms
The SIP standard, as specified in RFC 3261 , includes several security
mechanisms:
• S/MIME: Because SIP is using MIME for message bodies, S/MIME can be
used to send authenticated and encrypted messages between user
agents.
• Digest Authentication: SIP entities sharing a secret (e.g. a password) can
authenticate each other with a challenge-response mechanism.
• TLS & IPSec: Hop-by-hop security for SIP signaling can be achieved either
on the transport layer (TLS) or on the network layer (IP sec).
28. SIP-Secure over TLS
• SIPS is like HTTPS: Is set on top of
TCP only
• Signaling over sips URI:
sips:user@example.de;transport=tc
p, Demands for TLS along the
(signaling)path.
• Server authentication via Certificate
• Client authentication (mostly) via
username/digest.
• Client authentication via Certificate
possible
• Only Hop by Hop Security
• S/MIME − secure SDP
• Data format based on S/MIME mail.
• Encryption of the SDP portion of the
SIP message
• End-to-End or Hop by Hop allowed:
Tunneled (and S/MIME encrypted)
SDP also allowed
• Supports UDP or TCP: TCP is
recommended because of UDP
fragmentation.
S/MIME − secure SDP
29. CONCLUSION
The SIP is such a protocol, which does not have any built-in security.
This makes it more vulnerable to common VoIP attacks. In this
implementation of the SIP security threats and
countermeasures, the SIP secure model is designed to provide
security mechanisms by following the best practices for securing a
SIP based VOIP system.
30. CONCLUSION
The intention of this paper has been to present an overview of
important challenges and current activities on SIP security.
SIP is used to initiate IP Telephony communications. Thus, SIP
security will remain an active and interesting research area in the
near future.
31. THANK YOU
Muhammad Yeasir Arafat
Systems Engineer
Email: yeasir@dhakacom.com
yeasir08@yahoo.com
Dhakacom Limited
Dhaka, Bangladesh