WebRTC Integration from Tim Panton
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WebRTC Integration presentation from Tim Panton, given at WebRTC pre-conference workshop at Rich Communications.
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Westhawk integration
This document discusses various options for integrating WebRTC with carrier IMS networks. A gateway is currently needed due to incompatibilities between WebRTC and IMS protocols and codecs. Possible integration architectures described include using SIP in the browser with an SBC, REST calls to a WebRTC gateway, and XMPP through a gateway. Issues around identity mapping, scalability of gateways, and the immaturity of WebRTC on mobile are also covered.
Integrate WebRTC Voice / Video Call App Using Contus Fly
WebRTC has taken over internet calls, making it the onus of communication. WebRTC enables web application and mobile applications to stream peer-to-peer audio/video calls directly without the requirement of third-party integration.
Though there are multiple other languages to build WebRTC, such as peer.js, node.js, in Linux and Firebase, JavaScript rules out all the other programs with its phenomenal features. Let us take a close look at the parameters that pushes JavaScript over other languages.
Every other webRTC signaling server has the potential to build a cross-platform chat app that works on android/iOS. The short span of time JavaScript/node.js consumes to design a multi-user real-time application holds high potential to synchronize with the requirements the client delegates.
WebRTC video chat apps can be loaded as a server-side proxy and offers non-blocking IO, a set of features that can handle towering numbers of connections simultaneously. The system will not occupy excessive RAM space with video calls allowing a large number of calls to take place on the go.
Creating a video chat app using android/iOS javascript does not stop with providing video-voice internet calls through apps. The community is also growing up to provide exceptional modules that are absolutely ideal for integration of video call into the website and mobile applications. Socket.io is used to manage the constant communication between both the servers to provide real-time updates on the go.
Apart from building WebRTC for video chat app, opting for a third party integration of video call into website/app and video calling integration providers has the potential to create a peerless video chat app that runs on Android/iOS or Web - any platform of your choice.
Contus Fly is a readymade messaging SDK for Android & iOS, Chat API for Website, WebRTC solution provider, that has over a decade of experience in building chat apps. They have successfully delivered real-time video/voice chat functionality across multiple browsers & platforms for giant telecom services. The WebRTC signaling customization option they offer attract multinational brands as the video/voice call chat functionality operates with zero lag.
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A jQuery for WebRTC
WebRTC enables context based, embedded communication in any app or website. Skylink makes using WebRTC as simple as using jQuery for web developers.
Here is the link to the JS Remote Conf talk this presentation was held first: https://www.youtube.com/watch?v=x2IHJBp2TTo
SignalR Intro + WPDev
This document introduces SignalR, an open source library for ASP.NET that enables real-time web functionality. SignalR allows for persistent connections and events between client and server. It abstracts away the transport layer so applications can focus on two-way communication instead of polling. Live demos show how clients can call servers and vice versa without polling. SignalR works natively in .NET and supports platforms like iOS and Mono. While other technologies solve this problem, SignalR offers simplicity, ease of use within .NET, and support for WebSockets.
WebRTC: players, business models and implications for telecommunication carriers
- WebRTC provides real-time communication capabilities directly within web browsers using HTML5, with no plugins required. It is an open source technology backed by Google, Mozilla, and others.
- WebRTC uses common web technologies like JavaScript to enable rich media applications such as video chat and calling directly in the browser. However, signalling and network infrastructure are not defined.
- While the technology offers potential for innovative new services, many questions remain around business models and how existing players in telecommunications and over-the-top communication might be affected.
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Integrate WebRTC Voice / Video Call App Using Contus Fly
WebRTC has taken over internet calls, making it the onus of communication. WebRTC enables web application and mobile applications to stream peer-to-peer audio/video calls directly without the requirement of third-party integration.
Though there are multiple other languages to build WebRTC, such as peer.js, node.js, in Linux and Firebase, JavaScript rules out all the other programs with its phenomenal features. Let us take a close look at the parameters that pushes JavaScript over other languages.
Every other webRTC signaling server has the potential to build a cross-platform chat app that works on android/iOS. The short span of time JavaScript/node.js consumes to design a multi-user real-time application holds high potential to synchronize with the requirements the client delegates.
WebRTC video chat apps can be loaded as a server-side proxy and offers non-blocking IO, a set of features that can handle towering numbers of connections simultaneously. The system will not occupy excessive RAM space with video calls allowing a large number of calls to take place on the go.
Creating a video chat app using android/iOS javascript does not stop with providing video-voice internet calls through apps. The community is also growing up to provide exceptional modules that are absolutely ideal for integration of video call into the website and mobile applications. Socket.io is used to manage the constant communication between both the servers to provide real-time updates on the go.
Apart from building WebRTC for video chat app, opting for a third party integration of video call into website/app and video calling integration providers has the potential to create a peerless video chat app that runs on Android/iOS or Web - any platform of your choice.
Contus Fly is a readymade messaging SDK for Android & iOS, Chat API for Website, WebRTC solution provider, that has over a decade of experience in building chat apps. They have successfully delivered real-time video/voice chat functionality across multiple browsers & platforms for giant telecom services. The WebRTC signaling customization option they offer attract multinational brands as the video/voice call chat functionality operates with zero lag.
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This document discusses the rise of peer-to-peer communication on the web through WebRTC. WebRTC allows real-time communication like audio and video calls directly in web applications without plugins. It consists of APIs for getting media, creating peer connections, and exchanging data between peers. WebRTC enables embedded contextual communication across different applications on various platforms, though support is still limited on some browsers.
A jQuery for WebRTC
WebRTC enables context based, embedded communication in any app or website. Skylink makes using WebRTC as simple as using jQuery for web developers.
Here is the link to the JS Remote Conf talk this presentation was held first: https://www.youtube.com/watch?v=x2IHJBp2TTo
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This document introduces SignalR, an open source library for ASP.NET that enables real-time web functionality. SignalR allows for persistent connections and events between client and server. It abstracts away the transport layer so applications can focus on two-way communication instead of polling. Live demos show how clients can call servers and vice versa without polling. SignalR works natively in .NET and supports platforms like iOS and Mono. While other technologies solve this problem, SignalR offers simplicity, ease of use within .NET, and support for WebSockets.
WebRTC: players, business models and implications for telecommunication carriers
- WebRTC provides real-time communication capabilities directly within web browsers using HTML5, with no plugins required. It is an open source technology backed by Google, Mozilla, and others.
- WebRTC uses common web technologies like JavaScript to enable rich media applications such as video chat and calling directly in the browser. However, signalling and network infrastructure are not defined.
- While the technology offers potential for innovative new services, many questions remain around business models and how existing players in telecommunications and over-the-top communication might be affected.
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WebRTC is a free, open project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. It was released by Google in 2011 and it is becoming more famous day by day.
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WebRTC allows real-time communication capabilities like audio and video chat to be embedded directly into web pages without requiring plugins. It uses JavaScript APIs to handle communication between browsers over UDP connections. Key features include audio/video calls, file sharing, and integrating with other networks. Communication works using an offer/answer model where one peer sends an offer that is routed through a signaling server to the other peer, which responds with an answer.
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2. HTML5 aims to simplify development with new semantic elements, easier form handling, and making audio/video native elements.
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This document provides an overview of WebRTC in 3 parts:
1) What is WebRTC? WebRTC offers real-time communication directly in web browsers using JavaScript APIs and supports media codecs like VP8.
2) Entities in WebRTC including the browser, signaling techniques like WebSocket and XMPP, and protocols like STUN and TURN for NAT traversal.
3) How to learn WebRTC including recommended books, websites, and weekly newsletters that provide tutorials, code samples, and discussions around advances in the technology.
WebRTC - a History Lesson
WebRTC has progressed significantly in its first 3 years, moving from early experiments and proof of concepts to widespread adoption in browsers and innovative business applications. It started as an open source project at Google in 2011 and is now both an open standard specification and software stack. Major browsers like Chrome, Firefox, and Opera now support WebRTC natively. While adoption started with video chat apps, the technology is now used in verticals like education, healthcare, and more. Over 600 projects from vendors use WebRTC. In the next few years, the technology will continue transitioning to broader use in cloud services and reinventing communications with support from more players like Microsoft.
Hello 1 2 3, can you see me now?
Learning from the mistakes of the past and knowing where we stand at present will help us build the Internet video communication systems of the future. I present my point of view on the evolution, challenges and mistakes of the past, and, moving forward, describe the challenges in bridging the gap between web and VoIP. I highlight my contributions at various stages in the journey of Internet audio/video communication protocols.
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WebRTC allows direct peer-to-peer communication between browsers without plugins. It uses technologies like DTLS and SRTP for secure media, and ICE and TURN for network traversal through NATs and firewalls. The WebRTC API defines a JavaScript API and SDP standard for applications to establish sessions between peers. While WebRTC does not specify a signaling protocol, it is commonly used with SIP via gateways to connect to SIP networks and devices. WebRTC has many applications beyond just calls, including games, dating sites, and transferring arbitrary data directly in browsers.
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The document provides an overview of WebRTC, including:
- WebRTC allows real-time communication via voice, video and data sharing directly in web browsers.
- It has been standardized by the IETF RTCWEB working group and W3C.
- Key components include the JavaScript API, ICE for firewall/NAT traversal, DTLS-SRTP for encryption, and codecs like Opus and VP8.
- WebRTC is implemented in browsers like Chrome and Firefox but compatibility and which video codec to mandate are still issues.
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Baby Steps: A WebRTC Tutorial
This document provides an overview of WebRTC in 3 parts:
1) What is WebRTC? WebRTC offers real-time communication directly in web browsers using JavaScript APIs and supports media codecs like VP8.
2) Entities in WebRTC including the browser, signaling techniques like WebSocket and XMPP, and protocols like STUN and TURN for NAT traversal.
3) How to learn WebRTC including recommended books, websites, and weekly newsletters that provide tutorials, code samples, and discussions around advances in the technology.
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WebRTC has progressed significantly in its first 3 years, moving from early experiments and proof of concepts to widespread adoption in browsers and innovative business applications. It started as an open source project at Google in 2011 and is now both an open standard specification and software stack. Major browsers like Chrome, Firefox, and Opera now support WebRTC natively. While adoption started with video chat apps, the technology is now used in verticals like education, healthcare, and more. Over 600 projects from vendors use WebRTC. In the next few years, the technology will continue transitioning to broader use in cloud services and reinventing communications with support from more players like Microsoft.
Hello 1 2 3, can you see me now?
Learning from the mistakes of the past and knowing where we stand at present will help us build the Internet video communication systems of the future. I present my point of view on the evolution, challenges and mistakes of the past, and, moving forward, describe the challenges in bridging the gap between web and VoIP. I highlight my contributions at various stages in the journey of Internet audio/video communication protocols.
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It is targeted at bringing people up to speed with what WebRTC is, how people and vendors are using it today and placing it also in the context of the telecom world (which is the focus of this specific conference).
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WebRTC allows direct peer-to-peer communication between browsers without plugins. It uses technologies like DTLS and SRTP for secure media, and ICE and TURN for network traversal through NATs and firewalls. The WebRTC API defines a JavaScript API and SDP standard for applications to establish sessions between peers. While WebRTC does not specify a signaling protocol, it is commonly used with SIP via gateways to connect to SIP networks and devices. WebRTC has many applications beyond just calls, including games, dating sites, and transferring arbitrary data directly in browsers.
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The document discusses Session Initiation Protocol (SIP), including its purpose for initiating and managing multimedia sessions over the Internet, the various SIP entities like user agents and servers, common SIP message types, how SIP establishes and terminates calls, call redirection and proxying, using SIP for instant messaging and Internet telephony, additional SIP applications, and the future of SIP. It provides an overview of SIP, describing its core functionality and some key aspects of how it works at a high level.
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Web Real Time Communication (WebRTC) is a new web standard that enables real-time communication directly in web browsers. It allows for peer-to-peer connections between browsers for video calling, file sharing, and other applications. WebRTC uses JavaScript APIs and HTML5 to access cameras and microphones, establish peer connections, and exchange streaming media and data without plugins. It provides encryption and security to ensure private communication.
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The document provides an overview of WebRTC, including:
- WebRTC allows real-time communication via voice, video and data sharing directly in web browsers.
- It has been standardized by the IETF RTCWEB working group and W3C.
- Key components include the JavaScript API, ICE for firewall/NAT traversal, DTLS-SRTP for encryption, and codecs like Opus and VP8.
- WebRTC is implemented in browsers like Chrome and Firefox but compatibility and which video codec to mandate are still issues.
Aarti Sip
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WebRTC Reborn - Cloud Expo / WebRTC Summit
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- WebRTC supports real-time communication of arbitrary data between browsers using data channels in addition to audio and video.
- Data channels use SCTP over DTLS for transport, providing reliability, security, and NAT traversal. They have a WebSocket-like API.
- Early experiments show potential use cases but also immature implementations and possible overkill for some scenarios compared to WebSockets.
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Demo application using XMPP for signalling
open source webRTC using websockets is here: implenentationhttps://github.com/pizuricv/webRTC-over-websockets
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My slide deck from the session I gave at Twilio's Signal event May 2015.
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The document describes a WebRTC gateway product that connects browser-based telephony using WebRTC standards to traditional VoIP networks and devices using SIP. The gateway allows users on any WebRTC-supported browser to make audio and video calls to SIP phones and networks, integrating browser communication into web applications without APIs or SDKs. It also provides security features like TLS encryption for calls between browsers and the gateway. The gateway can be quickly deployed on cloud platforms like Amazon Web Services.
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This document discusses the requirements, technologies, and design for building a live video broadcasting system using WebRTC. The key requirements are to enable one-way live video broadcasting with minimal noise and delay in real-time without plugins. It evaluates video and streaming codecs like H.264, VP8, VP9 and streaming protocols like RTMP, HLS, and WebRTC. It provides an overview of how WebRTC works using signaling, peer connection, and ICE protocols. It also includes diagrams illustrating the logical, process, deployment and user views of the proposed system using Node.js, HTML, Firebase and WebRTC to enable multi-user connectivity. The document discusses testing the system with up to 100 users on 10
Building a WebRTC Communication and collaboration platform - techleash barcamp
WebRTC allows for multimedia communication directly through the browser without plugins. It uses peer-to-peer connections for media and can utilize various signaling protocols to broker connections. WebRTC is supported by all major browsers and can be used for personal communication, telehealth, call centers, IoT, and other use cases. However, concerns remain around firewall traversal, security, integration with other standards, and legal issues regarding privacy and lawful interception.
WebRCT
WebRTC allows real-time communication directly between web browsers without plugins. It provides APIs for getting audio and video streams and exchanging data peer-to-peer. While WebRTC enables direct communication, it still requires a server for user discovery and signaling to handle network traversal issues. Current limitations include a need for cloud infrastructure, a lack of support for native apps, inefficiencies for multiparty calls, and an inability to record streams.
Webrt integration by altanai bisht
Explains WebRTC , 3 modes of WebRTC integration by telecom service provider , security concerns . Also throws light on untouched areas of WebRTC integration encountered in during actual project .
Webrtc in Real world
The document discusses WebRTC and its advantages for real-time communication over the web. WebRTC allows web browsers to communicate in real-time through simple JavaScript APIs without requiring plugins. It has a simple architecture using APIs for accessing cameras, connecting to other users, and transferring data like files. WebRTC provides an easy way to build video chat and other real-time applications using only a few lines of code. However, some browsers have not fully implemented WebRTC APIs yet.
Status of WebRTC across Asia by Alan Quayle +++
Status of WebRTC across Asia by Alan Quayle, and a group of leading experts contributing to the reality, not the hype, of WebRTC.
It’s 2020, WebRTC (Web Real Time Communications) became known in 2011 when Google open sourced intellectual property it had bought in previous years. Gossip about those acquisitions began in 2009. The IETF (Internet Engineering Task Force) was already laying the groundwork with Opus (voice codec) officially in 2010, and back in 2009 the discussion process started that became WebRTC. It’s been roughly one decade. Did WebRTC change everything? Is WebRTC everywhere?
WebRTC myths and misconceptions. Understanding the two components of WebRTC, the open source project, and the standards track.
Reviewing the achievements of WebRTC across Asia.
Understanding why ‘WebRTC’ companies such as Vidyo and Tokbox did not achieve big exits.
What is the current status of WebRTC, where are the standards, where is the innovation edge?
What is happening across Asia on WebRTC? Understanding the difference service providers adoption of WebRTC. Across telcos, CPaaS, UCaaS. CCaaS, in-app communication platforms, and enterprises.
Case studies on WebRTC implementation across Asia.
Recommendations for WebRTC in Asia.
Boost JBoss AS7 with HTML5 WebRTC for Real Time Communications
HTML5 WebRTC, for Web Real Time Communications is free, open secifications to enable rich, high quality, Real Time Communications applications to be developed in the browser via simple Javascript APIs and HTML5. Major browsers already support or will support it soon natively. This talk will present an overview of WebRTC, how it is already revolutionizing the Web and changing the Telco industry. A couple of emblematic use cases will be also explored to show the potential of WebRTC in different enterprise markets and a live demo of a 1 to 1 WebRTC Video Conference will also be performed followed by a detailed explanation on how it was achieved as well as what JBoss AS7 additions were required to make it work
SignalR + Mobile Possibilities
This document introduces SignalR, a library for building real-time web functionality. SignalR allows adding real-time web functionality to applications by abstracting away the transport layer and enabling features like server push notifications without polling. It works natively in .NET and supports cross-platform clients. Live demos are provided showing how SignalR allows seamless client-server communication without polling. SignalR is presented as a simpler alternative to existing technologies for real-time web applications.
Getting the best out of WebRTC
This document discusses best practices for deploying WebRTC to replace or augment existing SIP-based phone systems. It covers choosing appropriate codecs to balance bandwidth usage and call quality for different use cases. It also addresses WebRTC-specific considerations like ICE, DTLS, and asymmetric call patterns. Performance metrics are provided from test calls using different codecs on an Asterisk server. The presentation includes diagrams of common WebRTC deployment architectures and links to live demos.
Getting the Best Out Of WebRTC - Astricon 2014
This document discusses best practices for deploying WebRTC to replace or augment existing SIP-based phone systems. It covers choosing appropriate codecs to balance bandwidth usage and call quality for different use cases. It also addresses WebRTC-specific considerations like ICE, DTLS, and asymmetric call patterns. Performance metrics are provided from test calls using different codecs on an Asterisk server. The presentation includes diagrams of common WebRTC deployment architectures and links to live demos.
WebSphere Liberty Rtcomm: WebRTC Middleware for the Enterprise
In order to provide the type of services their customers crave, your clients need to be able to provide blazing fast communication capabilities and access important information in the blink of an eye. WebRTC (Web Real-Time Communications) allows for the creation of next-generation communication applications without the need for browser plugins. WebSphere Application Server Liberty Profile is changing the way people communicate by making it easy to provide web page context as part of real-time conversations. This webinar will cover all of the real-time communications features recently released in WebSphere Liberty, including the new Rtcomm feature for rapid development of WebRTC based applications, and the open-source Rtcomm client-side libraries. (link to webinar replay: http://www.websphereusergroup.org/khatch/go/gallery/item/1543395?type=video)
WebRTC, Whats in it for me?
You already have working infrastructure. You know the ins and outs of your video protocol.
Everything is working, but you feel like things could work even better. If so, this talk is for you!
This talk explores all the things WebRTC could unlock for you. There could be solutions for problems you didn't
even realize were fixable!
ITCamp 2011 - Florin Cardasim - Duplex Communications with WCF and Azure
This document summarizes an IT camp presentation on duplex communication with WCF and Azure. The presentation covered enterprise duplex communication using WCF bindings and a router service, as well as web duplex communication for browser clients using polling, comet/long polling, and WebSockets. It provided demos of connecting enterprises using NetTcpBinding, WsDualHttpBinding, a router service, and Azure Service Bus. It also demonstrated WebSockets communication and discussed server implementations in various languages.
C koehncke webinar v2
This document discusses WebRTC and how GENBAND's SPiDR software can help companies implement WebRTC applications. SPiDR provides all the necessary elements integrated into a start-up suite for WebRTC, including a REST interface to SIP infrastructure, WebRTC session control, media management for WebRTC, and security features. SPiDR allows companies to leverage their existing SIP investments to build WebRTC applications and reach customers through a web interface. Some specific ideas discussed for how SPiDR could add value include offering access to existing subscription services on multiple devices, extending SIP trunking capabilities to web browsers, providing free video calling services, and offering low-cost unified communications soft phone and portal experiences.
Boost JBoss AS7 with HTML5 WebRTC for Real Time Communications
WebRTC, for Web Real Time Communications is a free, open project to enable rich, high quality, Real Time Communications applications to be developed in the browser via simple Javascript APIs and HTML5. Major browsers already support or will support it soon natively. This talk will present an overview of WebRTC, how it is already revolutionizing the Web and changing the Telco industry. A couple of emblematic use cases will be also explored to show the potential of WebRTC in different enterprise markets and a live demo of a 1 to 1 WebRTC Video Conference will also be performed followed by a detailed explanation on how it was achieved as well as what JBoss AS7 additions were required to make it work
WebRTC Infrastructure Design
The document describes the architecture for a WebRTC infrastructure. It discusses using a signaling server to facilitate peer-to-peer connections between WebRTC clients by relaying session description protocol (SDP) messages. It also describes using TURN servers, STUN, and containers to handle load balancing. The document outlines considerations for security, performance monitoring, and testing the design.
The Enterprise wants WebRTC -- and it needs Middleware to get it! (IIT RTC Co...
WebRTC is finally cracking the enterprise market. Maturing standards and wider platform adoption are helping WebRTC to find its way into mission critical enterprise applications. Whether it\'s financials like American Express or smaller businesses looking for innovative ways to engage their customers, WebRTC is changing the way business views real-time communications. Conversational media is Big Data to the enterprise and extracting every ounce of insight from every customer interaction requires middleware that plays well with existing Systems of Engagement. Issues like enterprise application integration, federation, analytics and their related security models bring with it requirements that must be well understood to succeed in this market. This session will explore what middleware means to WebRTC and what you need to make it work both in the cloud or on premise.
Similar to WebRTC Integration from Tim Panton (20)
Building a WebRTC Communication and collaboration platform - techleash barcamp
Building a WebRTC Communication and collaboration platform - techleash barcamp
Boost JBoss AS7 with HTML5 WebRTC for Real Time Communications
Boost JBoss AS7 with HTML5 WebRTC for Real Time Communications
WebSphere Liberty Rtcomm: WebRTC Middleware for the Enterprise
WebSphere Liberty Rtcomm: WebRTC Middleware for the Enterprise
ITCamp 2011 - Florin Cardasim - Duplex Communications with WCF and Azure
ITCamp 2011 - Florin Cardasim - Duplex Communications with WCF and Azure
Boost JBoss AS7 with HTML5 WebRTC for Real Time Communications
Boost JBoss AS7 with HTML5 WebRTC for Real Time Communications
The Enterprise wants WebRTC -- and it needs Middleware to get it! (IIT RTC Co...
The Enterprise wants WebRTC -- and it needs Middleware to get it! (IIT RTC Co...
More from Alan Quayle
What is a vCon?
TADSummit 2022 8/9 Nov Aveiro Portugal
Welcome to vCon! The next leap forward in the programmable communications industry.
Thomas Howe, CTO STROLID
Slides and Video
Why do we need vCon?
What is vCon?
How is it being used today?
Where is vCon going?
Supercharging CPaaS Growth & Margins with Identity and Authentication, Aditya...
TADSummit 2022 8/9 Nov Aveiro Portugal
Supercharging CPaaS Growth & Margins with Identity and Authentication
Aditya Khurjekar, GM Prove Protocol
Mobile networks were designed for communication, yet commerce is driving most of the demand for mobile connectivity today
The growth segments in today’s digital economy benefit from CPaaS APIs for Identity verification, authentication, proofs & claims
Commerce-enabling CPaaS APIs rely on the intrinsic security of mobile network and devices
Deterministic (rather than probabilistic) authentication drastically reduces fraud, hence increases margins
The secure element in mobile devices has been under-utilized by carriers
FIDO standard presents a horizontal application opportunity for hardware based (deterministic) authentication
Authenticated ID verification is key to secure yet seamless digital onboarding, leading to financial inclusion & consumer protection
The needs of the new crypto-based (web3) economy can also be satisfied with smart CPaaS offerings that preserve anonymity/pseudonymity
The imminent ubiquity of eSIMs is timely to fight fraud in the increasingly sophisticated digital & crypto-enabled economy
It’s time for a purpose-built global payments network!
Building a sub-second virtual ThunderDome: Considerations for mass scale sub-...
Building a sub-second virtual ThunderDome: Considerations for mass scale sub-second production broadcasts
Jerod Venema, CEO and Co-Founder, LiveSwitch
In the throes of the pandemic, the WWE debuted its ThunderDome, a world-first, large-scale installation of high resolution LED screens that transformed empty seats into live-streamed fans who joined over video from around the world. Performers in the ring and TV audiences at home could see and hear these virtual fans in real-time. LiveSwitch was selected to develop and manage the ThunderDome’s cloud video infrastructure.
How to enable low-latency, live video streamed via the internet capable of fostering real-time engagement between performers and audiences on a massive scale.
Massive-scale latency challenges and how to overcome them.
Current and future uses of programmable communications for live fan engagement.
What makes a cellular IoT API great? Tobias Goebel
What makes a cellular IoT API great?
Tobias Goebel, Principal Product Marketing Manager, IoT, Twilio
Why IoT SIMs need an API in the first place
The core functions needed in a cellular IoT API: SIM activation and deactivation, SIM status queries, Network access configuration, Pulling billing information and usage records, Troubleshooting, Device reachability
What matters in a good API (any API)
10 tips and tricks for how to find a good IoT SIM with a strong API
eSIM as Root of Trust for IoT security, João Casal
This document discusses the role of eSIM in new IoT security services. It begins by providing background on eSIM and SIM technology. It then outlines several ways eSIM can enhance IoT security, including:
1) Enabling zero-touch authentication of IoT devices in third-party services by leveraging the proven authentication of SIMs in cellular networks.
2) Hardening data encryption using the eSIM as a root of trust by generating encryption keys within the secure element of the eSIM.
3) Potential future roles like integrating the eSIM with threat detection to trigger authorization actions, and increasing the robustness of remote attestation through eSIM cryptographic abilities.
The document argues
Architecting your WebRTC application for scalability, Arin Sime
This document discusses how to architect WebRTC applications for scalability. It begins by outlining some of the challenges in building scalable WebRTC apps. It then presents 4 approaches to building apps: 1) To the WebRTC standard, 2) Unbundled WebRTC, 3) Using open-source media servers, and 4) Using communications platform as a service (CPaaS). Each approach has tradeoffs around cost, difficulty, and features included. The document also discusses using selectice forwarding units or multipoint control units to scale apps and considers architectures using orchestration and containers. It concludes with recommendations around optimizations, load testing, and future technologies.
CPaaS Conversational Platforms and Conversational Customer Service – The Expe...
TADSummit 2022 8/9 Nov Aveiro Portugal
CPaaS Conversational Platforms and Conversational Customer Service – The Experience Gap”?
Ben Waymark, Chief Technology Officer, Webio.
CPaaS players are doing the low hanging, simple conversations via their conversational design and plug in’s to the messenger layer, but what are they really hoping to achieve, and should they be aimed at the developer community?
No-code low-code configurable conversational customer support have done really well by integrating with customer ticketing, and integrating other platforms into their workflows. Kustomer.com was bought for a billion, something is going right there.
Conversational experiences are becoming part of the digital customer experience. What does this look like and why might this be important for other companies to understand?
Programmable Testing for Programmable Telcos, Andreas Granig
Programmable Testing for Programmable Telcos
Andreas Granig, Founder & CEO at Sipfront
Advantages and Challenges of automating real-time communication testing
How real-time communication testing could actually be quite pleasant
Creative ways to use typical server-side applications like kamailio and rtpengine as test clients
The revival of sipp, and how you create test scenarios 20 years after its invention
“Just show me the curl command”
How to best maximize the conversation data stream for your business? Surbhi R...
TADSummit 2022 8/9 Nov Aveiro Portugal
How to best maximize the conversation data stream for your business?
Surbhi Rathore, CEO & Co-Founder, Symbl.ai
How do we go from building a scalable pipeline of conversation data that merges and correlates with other types of data in the business and helps us makes decisions and predictions that are informed by conversations?
We will talk about context, real-time aspects of understanding and how you can use this data combined with sales, marketing, HR, support and other existing analytics to understand behavior and adapt to what works best in each of these functions.
We will go deep into specific use case and customer stories that have adopted Symbl’s conversation understanding platform to drive this change in their organization and give concrete examples of where to start.
Latest Updates and Experiences in Launching Local Language Tools, Karel Bourgois
TADSummit 2022, 8/9 Nov Aveiro Portugal
Latest Updates and Experiences in Launching Local Language Tools
Karel Bourgois, Founder Voxist, President Le Voice Lab, Exec Director Slatch, Chapter Pilot France AI Hub
Experiences with launching our own speech-to-text (French and English, both HD and Telephony audio, real-time and asynchronous).
‘Implicit Knowledge Management’ solution: using our STT engine we are indexing and searching thousands of hours of video to find those that discuss specific topics or identify people that are experts on those topics.
Latest updates on Voxist and its evolution to a “callbot.”
What Everyone Needs to Know about Protecting the CPaaS Ecosystem from Unlawfu...
This document discusses unlawful robocalls and solutions for CPaaS providers. It outlines that STIR/SHAKEN helps with some caller ID spoofing but not all, and leased phone numbers present challenges. It recommends CPaaS providers monitor customer usage of provided phone numbers and investigate customers' phone number reputations. The document introduces YouMail Score and Watch solutions that help identify unlawful calls and monitor phone number behaviors to improve call screening. It emphasizes that content-based call screening provides a virtually zero false positive rate compared to event-based screening alone.
Master the Audience Experience Multiverse: AX Best Practices and Success Stor...
Master the Audience Experience Multiverse: AX Best Practices and Success Stories
Ken Herron, Chief Growth Officer, UIB
Customers need you to help them solve their #1 problem – Audience Experience (AX).
Customers struggle with managing their differentiated brand journeys at scale in a post-pandemic world where their external and internal audiences decide the platforms, channels, and languages.
This session will share AX best practices and success stories from Europe, the Middle East, Asia, and the US for how enterprise and small business customers can control their respective brands, journeys, and audiences with a single brand voice –
Create/Control a differentiated AX
Respond in real-time
Mirror channels
Curate audiences
Secure conversational data
Monetize engagement
Scale monitoring
This session will include a live, interactive demo.
Open Source Telecom Software Survey 2022, Alan Quayle
The survey gathered responses from 120 participants in 2022 compared to 114 in 2021. It covered general topics such as DDoS attacks, security practices, STIR/SHAKEN implementation, IPv6 deployment, and expectations for major tech companies over the next two years. Key findings included that around half of participants have experienced a DDoS attack in 2022 and application-level attacks were as common as volumetric attacks. Most organizations take both reactive and proactive security approaches but more so reactive. STIR/SHAKEN implementation is ongoing with international carriers needing it to terminate traffic in North America. IPv6 deployment remains steady with no major differences between regions. This survey provides insights into trends in the open source tele
OpenSIPS 3.3 – Messaging in the IMS and UC ecosystems. Bogdan-Andrei Iancu
TADSummit 2022, 8/9 Nov Aveiro Portugal
OpenSIPS 3.3 – Messaging in the IMS and UC ecosystems.
Bogdan-Andrei Iancu, Founder and Developer at OpenSIPS Project
SIP also supports instant messaging and presence.
Review of Messaging in IMS
Review of Messaging in Unified Communications
OpenSIPS 3.3 in the messaging ecosystem
Review of implementation using Message Session Relay Protocol (RFC 4975, RFC 4976), groups multiple messages in sessions.
Conclusions: OpenSIPS 3.3 targets to implement various components of the overall SIP Instant MESSAGING ecosystem, from gateways and transport to services.
TADS 2022 - Shifting from Voice to Workflow Management, Filipe Leitao
TADSummit 2022, 8/9 Nov Aveiro Portugal
Shifting from Voice to Workflow Management
Filipe Leitão, Global Service Provider Channel SE, RingCentral
There is an ongoing consolidation of the Cloud Communications market where mainstream providers compete against each other for the same spaces, UCaaS / CCaaS / CPaaS.
Weapons of choice are the same for everyone: instant messaging, and audio & video conferencing. Most capabilities provided by mainstream UC providers are table stakes.
Find out how RingCentral is looking at UC from more than just a siloed perspective by going one step further and co-innovating with Service and Technology Providers to become a workflow management platform.
What happened since we last met TADSummit 2022, Alan Quayle
TADSummit 2022, 8/9 Nov Aveiro Portugal
What happened since we last met? Where is the Programmable Comms market going?
Alan Quayle, independent
3 years in Programmable Communications: 2020, 2021, and 2022 all done in 16 slides
Pandemic Consolidation
Post-pandemic Reckoning – I did predict what we’re seeing with Avaya
The Coming of Cost Competition
Messaging, will A2P SMS growth ever stop?
What’s the recession going to do to us?
The Voice AI Reckoning
After all the consolidation, where next? Twilio’s heading there – it’s about the data
And a few more predictions that are usually too optimistic
Stacuity - TAD Summit 2022 - Time to ditch the dumb-pipe, Mike Bromwich
TADSummit 2022, 8/9 Nov Aveiro Portugal
Time to ditch the ‘dumb-pipe’ – reinventing the core mobile network, to put developers first.
Mike Bromwich, CEO / Co-Founder Stacuity & Tim Dowling, Co-Founder Stacuity
The emergence of public cloud has revolutionized the way developers can muster and deploy virtual infrastructures, as and when required.
In contrast, mobile networks are still rigidly defined and protected by operators, who are unable or unwilling to offer such control and flexibility.
As a result, the mobile network operates as little more than a dumb-pipe (unless you have lots of patience and deep pockets).
Addressing this problem requires a different approach, not just the creation of a thin façade over legacy network elements.
How Stacuity is reinventing the core mobile network, to put developers first.
AWA – a Telco bootstrapping product development: Challenges with dynamic mark...
This document discusses AWA Network, a company that operates a CPaaS proxy providing APIs for SMS delivery. It outlines AWA's features, including anonymous resources, traffic simulation, provider management, pricing, routing, and SMS sending APIs. The document also presents three business models for AWA Network: 1) operating as a managed CPaaS proxy service, 2) allowing users to proxy their own CPaaS providers, and 3) enabling corporations to use AWA's infrastructure. It reflects on the challenges of scaling and finding a viable business and operations model.
Founding a Startup in Telecoms. The good, the bad and the ugly. João Camarate
TADSummit 2022 8/9 Nov Aveiro Portugal
Founding a Startup in Telecoms. The good, the bad and the ugly.
João Camarate, CTO at Broadvoice & GoContact.
A deep dive into the challenges and opportunities of starting a new venture in the telecom space while leveraging open-source
How to bring down your own RTC platform. Sandro Gauci
Sandro Gauci provides a walkthrough for performing distributed denial of service (DDoS) simulations on real-time communication (RTC) platforms to test security. He recommends starting with simple bandwidth saturation or protocol attacks before moving to specific application attacks. Tools are needed to distribute attacks from nodes, monitor systems, and shut down attacks. Findings should be analyzed with engineers through root cause analysis and documented. Solutions may include updates, rate limiting, or code changes. Regular testing ensures a more robust RTC platform.
More from Alan Quayle (20)
Supercharging CPaaS Growth & Margins with Identity and Authentication, Aditya...
Supercharging CPaaS Growth & Margins with Identity and Authentication, Aditya...
Building a sub-second virtual ThunderDome: Considerations for mass scale sub-...
Building a sub-second virtual ThunderDome: Considerations for mass scale sub-...
What makes a cellular IoT API great? Tobias Goebel
What makes a cellular IoT API great? Tobias Goebel
eSIM as Root of Trust for IoT security, João Casal
eSIM as Root of Trust for IoT security, João Casal
Architecting your WebRTC application for scalability, Arin Sime
Architecting your WebRTC application for scalability, Arin Sime
CPaaS Conversational Platforms and Conversational Customer Service – The Expe...
CPaaS Conversational Platforms and Conversational Customer Service – The Expe...
Programmable Testing for Programmable Telcos, Andreas Granig
Programmable Testing for Programmable Telcos, Andreas Granig
How to best maximize the conversation data stream for your business? Surbhi R...
How to best maximize the conversation data stream for your business? Surbhi R...
Latest Updates and Experiences in Launching Local Language Tools, Karel Bourgois
Latest Updates and Experiences in Launching Local Language Tools, Karel Bourgois
What Everyone Needs to Know about Protecting the CPaaS Ecosystem from Unlawfu...
What Everyone Needs to Know about Protecting the CPaaS Ecosystem from Unlawfu...
Master the Audience Experience Multiverse: AX Best Practices and Success Stor...
Master the Audience Experience Multiverse: AX Best Practices and Success Stor...
Open Source Telecom Software Survey 2022, Alan Quayle
Open Source Telecom Software Survey 2022, Alan Quayle
OpenSIPS 3.3 – Messaging in the IMS and UC ecosystems. Bogdan-Andrei Iancu
OpenSIPS 3.3 – Messaging in the IMS and UC ecosystems. Bogdan-Andrei Iancu
TADS 2022 - Shifting from Voice to Workflow Management, Filipe Leitao
TADS 2022 - Shifting from Voice to Workflow Management, Filipe Leitao
What happened since we last met TADSummit 2022, Alan Quayle
What happened since we last met TADSummit 2022, Alan Quayle
Stacuity - TAD Summit 2022 - Time to ditch the dumb-pipe, Mike Bromwich
Stacuity - TAD Summit 2022 - Time to ditch the dumb-pipe, Mike Bromwich
AWA – a Telco bootstrapping product development: Challenges with dynamic mark...
AWA – a Telco bootstrapping product development: Challenges with dynamic mark...
Founding a Startup in Telecoms. The good, the bad and the ugly. João Camarate
Founding a Startup in Telecoms. The good, the bad and the ugly. João Camarate
How to bring down your own RTC platform. Sandro Gauci
How to bring down your own RTC platform. Sandro Gauci
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Cosa hanno in comune un mattoncino Lego e la backdoor XZ?
ABSTRACT: A prima vista, un mattoncino Lego e la backdoor XZ potrebbero avere in comune il fatto di essere entrambi blocchi di costruzione, o dipendenze di progetti creativi e software. La realtà è che un mattoncino Lego e il caso della backdoor XZ hanno molto di più di tutto ciò in comune.
Partecipate alla presentazione per immergervi in una storia di interoperabilità, standard e formati aperti, per poi discutere del ruolo importante che i contributori hanno in una comunità open source sostenibile.
BIO: Sostenitrice del software libero e dei formati standard e aperti. È stata un membro attivo dei progetti Fedora e openSUSE e ha co-fondato l'Associazione LibreItalia dove è stata coinvolta in diversi eventi, migrazioni e formazione relativi a LibreOffice. In precedenza ha lavorato a migrazioni e corsi di formazione su LibreOffice per diverse amministrazioni pubbliche e privati. Da gennaio 2020 lavora in SUSE come Software Release Engineer per Uyuni e SUSE Manager e quando non segue la sua passione per i computer e per Geeko coltiva la sua curiosità per l'astronomia (da cui deriva il suo nickname deneb_alpha).
みなさんこんにちはこれ何文字まで入るの？40文字以下不可とか本当に意味わからないけどこれ限界文字数書いてないからマジでやばい文字数いけるんじゃないの？えこ...
ここ3000字までしか入らないけどタイトルの方がたくさん文字入ると思います。
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WebRTC Integration from Tim Panton
- 1. WEBRTC INTEGRATION OPTIONS Tim Panton, Westhawk Ltd – @steely_glint
- 2. Demo – call a mobile Demo calls my mobile from a browser
- 3. Not everyone wants to interop Games Dating sites Whiteboards OTT Mayday For these sites a home grown signaling protocol may be simplest/best. (highest value apps will be in this class)
- 4. Problem statement WebRTC HTTP(S) transport RIA 2.0 media Encrypted Opus VP8 ? Carrier IMS SIP RTP Cleartext Ulaw (729, amr?) H263/4
- 5. Gateway needed ? Unless and until those incompatibilities go away.
- 6. HTTP to SIP – SIP in the Browser Use javascript to build SIP messages and protocol Wrap in HTTP (or Web-sockets) Send to webserver Webserver unwraps and forwards to IMS
- 7. SIP in the browser Browser JS SIP SIP in HTTP WebSocket Server UDP SIP IMS
- 8. Problems You still have a gateway – albeit a thin one. You have javascript injecting SIP messages into IMS The SDP isn’t compatible The media isn’t compatible What is Early media in a browser ? You have your SIP credentials out on the internet.
- 9. SIP in the browser with SBC DMZ Browser JS SIP SIP in HTTP WebSocket Server UDP SIP SBC UDP SIP IMS
- 10. Remaining Problems The media isn’t compatible What is Early media in a browser ? You have SIP credentials out on the internet.
- 11. SIP in the browser with SBC, Media Gateway and Registration proxy DMZ Browser JS SIP SIP in HTTP RIA 2.0 WebSocket Server UDP SIP Proxy Reg SBC Media GW UDP SIP IMS RTP ulaw
- 12. REST in the browser Use web ‘RESTful’ commands Sent from the browser To a webRTC gateway Gateway generates the SIP IMS needs Gateway controls transcode resource
- 13. REST in the browser with Gateway DMZ Browser app REST/HTTP WebRTC gateway UDP SIP IMS
- 14. Problems Need to map from web Identity to SIP Select a web identity provider webRTC gateways don’t scale (yet) No standard for REST messages Home rolled protocol (may have holes)
- 15. XMPP in the browser with SBC DMZ Browser app XMPP/BOSH/H TTP WebRTC gateway UDP SIP IMS
- 16. Problems Need to map from web Identity to SIP Select a web identity provider webRTC gateways don’t scale (yet) More complex than necessary Needless protocol mapping? However BOSH is tested XMPP well defined and federates
- 17. Did we forget mobile? WebRTC isn’t mobile first yet.
- 18. WebRTC on Mobile Browser isn’t a natural interface WebRTC codecs are heavy on battery No native App friendly API (yet) SIP (if used) not an efficient mobile protocol Audio hardware on android variable Both Chrome and firefox on Android support webRTC Expect to see RIA 2.0 with native APIs
- 19. Identity. DTLS can carry a certificate – but which one should be used?
- 20. Multiple identities on the web When I call from a webpage, which identity do I want to present? E164 to the shop Facebook Id to my fb friends Anon to the game Pseudo id to dating site Do I ever want to present facebook ID to G+ users?
- 21. Demo Phono using a Jira identity. (perhaps)