The document provides an overview of three main voice over IP protocols: SIP, H.323, and MGCP. It describes the architecture and components of SIP and H.323, including user agents, proxies, registrars, and gateways. It also discusses how SIP and H.323 establish communication sessions and handle registration, call setup and teardown. MGCP is mentioned but not described in detail.
Brief introduction into SIP protocol, how it works, common problems to solve. Tech. details about handshake, SIP Trunks and SIP trunking. Market research.
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.
Brief introduction into SIP protocol, how it works, common problems to solve. Tech. details about handshake, SIP Trunks and SIP trunking. Market research.
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.
Session Initiation Protocol is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
Short presentation that introduces the basic concepts of SIP.
Note, some of the slides requires animations and will therefore look strange when viewed without them (e.g., see slide 17). Email me if you want the full presentation and I'll send it to you...
One of the most widely observed fraud scenarios is the case of a malicious user detecting the address of a PSTN gateway and accessing that gateway directly. Once the attacker has managed to access the gateway the attacker can start selling telephony minutes through that
gateway.
The ABC SBC establishes a secure border between the VoIP service provider’s core VoIP components–e.g., PSTN gateway, SIP proxy and application servers- and the subscribers. As the border element, the ABC SBC hides the details of the operator’s network from subscribers
and absorbs any attacks and sudden spikes in the subscriber traffic. Further, the mediation features of the ABC SBC shield the operator’s network from malfunctioning user agents and any interoperability issues.
The multi-functional IP-PBX delivers high performance. Simplified management, reduced communication cost, seamless connectivity with remote users and between geographically dispersed branches, advanced communication means and enhanced productivity are apparent benefits. The system employs open-standard SIP protocol and is hence interoperable with SIP proxies, gateways and IP phones. Communication of small and mid-sized enterprises as well as geographically distributed offices, remote workers and contact centre is much simplified and enhanced with SAPEX.
Current trends and innovations in voice over IPALTANAI BISHT
Learn how to implement an open-source webrtc Click to dial or VOIP setup for their enterprises and also the new innovative add-on tech available for a basic VOIP system such as auto-attendants.
VoIP vs Telecom Providers
SIP Servers types
Open-source tool and technologies in VOIP
Opensip
Kamailio
Freeswitch
Media Handling
Webrtc
Machine learning in VoIP
Call Classifier
Fraud Detector
NLP and Auto attendants
VoIP to telecom bridging
Session Initiation Protocol is a signaling communications protocol, widely used for controlling multimedia communication sessions such as voice and video calls over Internet Protocol (IP) networks.
Short presentation that introduces the basic concepts of SIP.
Note, some of the slides requires animations and will therefore look strange when viewed without them (e.g., see slide 17). Email me if you want the full presentation and I'll send it to you...
One of the most widely observed fraud scenarios is the case of a malicious user detecting the address of a PSTN gateway and accessing that gateway directly. Once the attacker has managed to access the gateway the attacker can start selling telephony minutes through that
gateway.
The ABC SBC establishes a secure border between the VoIP service provider’s core VoIP components–e.g., PSTN gateway, SIP proxy and application servers- and the subscribers. As the border element, the ABC SBC hides the details of the operator’s network from subscribers
and absorbs any attacks and sudden spikes in the subscriber traffic. Further, the mediation features of the ABC SBC shield the operator’s network from malfunctioning user agents and any interoperability issues.
The multi-functional IP-PBX delivers high performance. Simplified management, reduced communication cost, seamless connectivity with remote users and between geographically dispersed branches, advanced communication means and enhanced productivity are apparent benefits. The system employs open-standard SIP protocol and is hence interoperable with SIP proxies, gateways and IP phones. Communication of small and mid-sized enterprises as well as geographically distributed offices, remote workers and contact centre is much simplified and enhanced with SAPEX.
Current trends and innovations in voice over IPALTANAI BISHT
Learn how to implement an open-source webrtc Click to dial or VOIP setup for their enterprises and also the new innovative add-on tech available for a basic VOIP system such as auto-attendants.
VoIP vs Telecom Providers
SIP Servers types
Open-source tool and technologies in VOIP
Opensip
Kamailio
Freeswitch
Media Handling
Webrtc
Machine learning in VoIP
Call Classifier
Fraud Detector
NLP and Auto attendants
VoIP to telecom bridging
This presentation contain basic knowledge about how voIP work and what are the security threat in voIP. It will also contain how we can prevent attack on voIP system.
Presented by: Muhammad Yeasir Arafat
ElastixWorld 2013
Today, the session initiation protocol (SIP) is the predominant protocol for IP Telephony Signalling. The intention of this paper is to present an overview of IP Telephony security issues - both current and future – focusing on SIP. We start by presenting some fundamental differences between IP Telephony and the public switched telephone network (PSTN). We then look at specific problems for SIP signalling that arise from these differences. We summarize current activities regarding SIP security, including recent developments in the research community and standardization efforts within the IETF. Finally, the paper gives an outlook on the security issues of IP Telephony.
Download paper at: http://elx.ec/ssiipt
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.
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.
you can be friend with me on orkut
"mangalforyou@gmail.com" : i belive in sharing the knowledge so please send project reports ,seminar and ppt. to me .
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
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
Elevating Tactical DDD Patterns Through Object CalisthenicsDorra BARTAGUIZ
After immersing yourself in the blue book and its red counterpart, attending DDD-focused conferences, and applying tactical patterns, you're left with a crucial question: How do I ensure my design is effective? Tactical patterns within Domain-Driven Design (DDD) serve as guiding principles for creating clear and manageable domain models. However, achieving success with these patterns requires additional guidance. Interestingly, we've observed that a set of constraints initially designed for training purposes remarkably aligns with effective pattern implementation, offering a more ‘mechanical’ approach. Let's explore together how Object Calisthenics can elevate the design of your tactical DDD patterns, offering concrete help for those venturing into DDD for the first time!
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.
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
Slack (or Teams) Automation for Bonterra Impact Management (fka Social Soluti...Jeffrey Haguewood
Sidekick Solutions uses Bonterra Impact Management (fka Social Solutions Apricot) and automation solutions to integrate data for business workflows.
We believe integration and automation are essential to user experience and the promise of efficient work through technology. Automation is the critical ingredient to realizing that full vision. We develop integration products and services for Bonterra Case Management software to support the deployment of automations for a variety of use cases.
This video focuses on the notifications, alerts, and approval requests using Slack for Bonterra Impact Management. The solutions covered in this webinar can also be deployed for Microsoft Teams.
Interested in deploying notification automations for Bonterra Impact Management? Contact us at sales@sidekicksolutionsllc.com to discuss next steps.
Dev Dives: Train smarter, not harder – active learning and UiPath LLMs for do...UiPathCommunity
💥 Speed, accuracy, and scaling – discover the superpowers of GenAI in action with UiPath Document Understanding and Communications Mining™:
See how to accelerate model training and optimize model performance with active learning
Learn about the latest enhancements to out-of-the-box document processing – with little to no training required
Get an exclusive demo of the new family of UiPath LLMs – GenAI models specialized for processing different types of documents and messages
This is a hands-on session specifically designed for automation developers and AI enthusiasts seeking to enhance their knowledge in leveraging the latest intelligent document processing capabilities offered by UiPath.
Speakers:
👨🏫 Andras Palfi, Senior Product Manager, UiPath
👩🏫 Lenka Dulovicova, Product Program Manager, UiPath
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.
Transcript: Selling digital books in 2024: Insights from industry leaders - T...BookNet Canada
The publishing industry has been selling digital audiobooks and ebooks for over a decade and has found its groove. What’s changed? What has stayed the same? Where do we go from here? Join a group of leading sales peers from across the industry for a conversation about the lessons learned since the popularization of digital books, best practices, digital book supply chain management, and more.
Link to video recording: https://bnctechforum.ca/sessions/selling-digital-books-in-2024-insights-from-industry-leaders/
Presented by BookNet Canada on May 28, 2024, with support from the Department of Canadian Heritage.
Securing your Kubernetes cluster_ a step-by-step guide to success !KatiaHIMEUR1
Today, after several years of existence, an extremely active community and an ultra-dynamic ecosystem, Kubernetes has established itself as the de facto standard in container orchestration. Thanks to a wide range of managed services, it has never been so easy to set up a ready-to-use Kubernetes cluster.
However, this ease of use means that the subject of security in Kubernetes is often left for later, or even neglected. This exposes companies to significant risks.
In this talk, I'll show you step-by-step how to secure your Kubernetes cluster for greater peace of mind and reliability.
4. SIP, H.323 and MGCP RTCP RTP IP MGCP Call Control and Signaling Signaling and Gateway Control Media H.225 Q.931 H.323 H.323 Version 1 and 2 supports H.245 over TCP, Q.931 over TCP and RAS over UDP. H.323 Version 3 and 4 supports H.245 over UDP/TCP and Q.931 over UDP/TCP and RAS over UDP. SIP supports TCP and UDP. TCP RAS UDP SIP H.245 Audio/ Video RTSP
6. What is SIP? “ ” Session Initiation Protocol - An application layer signaling protocol that defines initiation, modification and termination of interactive, multimedia communication sessions between users. IETF RFC 2543 Session Initiation Protocol
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8. SIP Distributed Architecture User Agent Gateway PSTN SIP Components Redirect Server Location Server Registrar Server Proxy Server Proxy Server
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19. Simplified SIP Call Setup and Teardown 302 (Moved Temporarily) ACK INVITE 302 (Moved Temporarily) ACK Call Teardown Media Path Call Setup INVITE 200 (OK) 200 (OK) INVITE 180 (Ringing) 180 (Ringing) 180 (Ringing) 200 (OK) ACK ACK ACK RTP MEDIA PATH BYE BYE BYE 200 (OK) 200 (OK) 200 (OK) INVITE Location/Redirect Server Proxy Server Proxy Server User Agent User Agent INVITE
30. What is H.323? “ ” Describes terminals and other entities that provide multimedia communications services over Packet Based Networks (PBN) which may not provide a guaranteed Quality of Service. H.323 entities may provide real-time audio, video and/or data communications. ITU-T Recommendation H.323 Version 4
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32. H.323 Components Circuit Switched Networks Terminal Gateway Packet Based Networks Multipoint Control Unit Gatekeeper
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38. Other ITU H. Recommendation that work with H.323 Protocol Description H.235 Specifies security and encryption for H.323 and H.245 based terminals. H.450.N H.450.1 specifies framework for supplementary services. H.450.N recommendation specifies supplementary services such as call transfer, call diversion, call hold, call park, call waiting, message waiting indication, name identification, call completion, call offer, and call intrusion. H.246 Specifies internetworking of H Series terminals with circuit switched terminals.
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42. Versions of H.323 Version Reference for key feature summary H.323 Version 3 http://www.packetizer.com/iptel/h323/whatsnew_v3.html Date September 1999 H.323 Version 1 New release. Refer to the specification. http://www. packetizer .com/ iptel /h323/ May 1996 H.323 Version 2 http://www.packetizer.com/iptel/h323/whatsnew_v2.html January 1998 H.323 Version 4 November 2000 http://www.packetizer.com/iptel/h323/whatsnew_v4.html
47. Table 1 - SIP and H.323 H.323 SIP ITU. IETF. Peer-to-Peer. Peer-to-Peer. Telephony based. Borrows call signaling protocol from ISDN Q.SIG. Internet based and web centric. Borrows syntax and messages from HTTP. Intelligent H.323 terminals. Intelligent user agents. H.323 Gatekeeper. SIP proxy, redirect, location, and registration servers. IMTC sponsors interoperability events among SIP, H.323, and MGCP. For more information, visit: http://www.imtc.org/ Widespread. Interoperability testing between various vendor’s products is ongoing at SIP bakeoffs. SIP is gaining interest. Information Standards Body Relationship Origins Client Core servers Current Deployment Interoperability
48. Table 2 - SIP and H.323 Information H.323 SIP Capabilities Exchange Supported by H.245 protocol. H.245 provides structure for detailed and precise information on terminal capabilities. SIP uses SDP protocol for capabilities exchange. SIP does not provide as extensive capabilities exchange as H.323. Control Channel Encoding Type Binary ASN.1 PER encoding. Text based UTF-8 encoding. Server Processing Version 1 or 2 – Stateful. Version 3 or 4 – Stateless or stateful. Stateless or stateful. Quality of Service Bandwidth management/control and admission control is managed by the H.323 gatekeeper. The H323 specification recommends using RSVP for resource reservation. SIP relies on other protocols such as RSVP, COPS, OSP to implement or enforce quality of service.
49. Table 3 - SIP and H.323 Information H.323 SIP Security Registration - If a gatekeeper is present, endpoints register and request admission with the gatekeeper. Authentication and Encryption - H.235 provides recommendations for authentication and encryption in H.323 systems. Registration - User agent registers with a proxy server. Authentication - User agent authentication uses HTTP digest or basic authentication. Encryption - The SIP RFC defines three methods of encryption for data privacy. Endpoint Location and Call Routing Uses E.164 or H323ID alias and a address mapping mechanism if gatekeepers are present in the H.323 system. Gatekeeper provides routing information. Uses SIP URL for addressing. Redirect or location servers provide routing information.
50. Table 4 – SIP and H.323 Information H.323 SIP Note: Basic call features include: call hold, call waiting, call transfer, call forwarding, caller identification, and call park. Features Basic call features. Basic call features. Conferencing Basic conferencing without conference or floor control. Comprehensive audiovisual conferencing support. Data conferencing or collaboration defined by T.120 specification. Service or Feature Creation Supports flexible and intuitive feature creation with SIP using SIP-CGI and CPL. Some example features include presence, unified messaging, or find me/follow me. H.450.1 defines a framework for supplementary service creation.
53. What is MGCP? “ ” Media Gateway Control Protocol - A protocol for controlling telephony gateways from external call control elements called media gateway controllers or call agents. IETF RFC 2705 Media Gateway Control Protocol
H.323 – packet based multimedia communication system H.225 – call signaling protocol H.245 – call control protocol RAS – Registration Admission Signaling SIP – Session Initiation Protocol (RFC 2543) MGCP - Media Gateway Control Protocol H.248/Megaco – Media Gateway Control Protocol RTP – Real Time Transport Protocol (RFC 1889) RTCP – Real Time Transport Control Protocol (RFC 1889) RTSP – Real Time Streaming Protocol (RFC2324) UDP – User Datagram Protocol TCP - Transmission Control Protocol IP – Internet Protocol