This document reports on the results of a technology scout investigating a WebRTC to SIP gateway. It identifies challenges around transferring protocols, codecs, encryption and ICE between WebRTC and SIP. Possible open source candidates for the gateway are discussed, including FreeSWITCH, Doubango webrtc2sip, Janus and Kamailio with rtpengine. An overview of a potential WebRTC to SIP gateway architecture is provided. Limitations, performance, scalability and security considerations are also examined.
Tf web rtc-berlin-mai2016-sa8t2 roadmap-janmeijerJan Meijer
Short presentations to the 4th GÉANT TF-WebRTC meeting in Berlin, 2+3 May 2016. About the results of the WebRTC task in the GN4-1 project. License: CC Attribution 4.0 International
This document outlines an educational course on audio and video over IP. The course covers IP networking fundamentals and standards including TCP/IP, OSI models, and SMPTE ST 2110. It also examines IP infrastructure, routing, timing issues, switching, compression techniques and case studies for broadcast facilities transitioning to IP. The document provides an in-depth outline of topics covered in each session, from IP basics to designing and integrating both hybrid and fully IP-based outside broadcast trucks. The goal is to educate on best practices for implementing audio and video over IP workflows and infrastructure.
The document summarizes a presentation by Victor Pascual Avila and Antón Román Portabales on WebRTC standards updates from November 2014. It discusses the current state of WebRTC standards including supported audio and video codecs, signaling protocols, and interoperability with legacy VoIP/IMS networks. It also covers ongoing discussions around topics like preferred video codecs and the development of WebRTC browser APIs.
WebRTC Workshop 2013 given at the IMS World ForumAlan Quayle
The document provides an agenda for a WebRTC workshop covering the following key points:
- The workshop will provide a deep technical and business overview of WebRTC through presentations and demonstrations.
- Attendees will learn about the current status of WebRTC standardization and implementations, and what capabilities may emerge over the next 1-2 years.
- The workshop includes sessions on technology details like APIs, media protocols, and interoperability, as well as implications for service providers, enterprises, and use cases.
- An afternoon demo session will provide hands-on experiences of WebRTC applications from various companies and allow networking among attendees.
This document provides an overview and update on WebRTC standards from December 2014. It discusses what WebRTC is, including that it is a browser-embedded media engine. It describes the various WebRTC standards covering signaling, media codecs and protocols. There is no single defined signaling method for WebRTC. The document also discusses topics like the video codec battle between H.264 and VP8, browser support for WebRTC, and interworking WebRTC with legacy VoIP/IMS deployments.
This document provides an overview of WebRTC standardization efforts in 3 sentences or less:
WebRTC standardization is being addressed by the W3C for browser APIs, the IETF for protocols, and other groups like the SIP Forum and 3GPP. There are still open issues around browser support, signaling, interworking with SIP, video codecs, and security. The document concludes by introducing the speaker, Victor Pascual, as a technology consultant who can be contacted for further discussion on WebRTC standardization topics.
Each WebRTC deployment is implementing its own proprietary signaling mechanism. WebRTC signaling and media is incompatible with existing VoIP deployments, requiring gateways to bridge the two. The WebRTC API specification is still evolving and may take different forms across implementations. Standards bodies are working to define signaling interoperability and bridge WebRTC with existing networks like SIP, IMS and videoconferencing systems.
WebRTC provides a standardized profile for real-time communication that enables interoperability between browsers without plugins. It defines client-side APIs for audio and video calling as well as other real-time communication capabilities. The WebRTC architecture includes the API, codecs, transport mechanisms like STUN and TURN, and network I/O that allow real-time apps to run directly in browsers. Signaling is required to establish connections between users, and the standardization of WebRTC aims to improve interoperability compared to proprietary solutions. However, interoperability is not always in the best interests of businesses. Ultimately, the API is more important than the underlying protocols it uses.
Tf web rtc-berlin-mai2016-sa8t2 roadmap-janmeijerJan Meijer
Short presentations to the 4th GÉANT TF-WebRTC meeting in Berlin, 2+3 May 2016. About the results of the WebRTC task in the GN4-1 project. License: CC Attribution 4.0 International
This document outlines an educational course on audio and video over IP. The course covers IP networking fundamentals and standards including TCP/IP, OSI models, and SMPTE ST 2110. It also examines IP infrastructure, routing, timing issues, switching, compression techniques and case studies for broadcast facilities transitioning to IP. The document provides an in-depth outline of topics covered in each session, from IP basics to designing and integrating both hybrid and fully IP-based outside broadcast trucks. The goal is to educate on best practices for implementing audio and video over IP workflows and infrastructure.
The document summarizes a presentation by Victor Pascual Avila and Antón Román Portabales on WebRTC standards updates from November 2014. It discusses the current state of WebRTC standards including supported audio and video codecs, signaling protocols, and interoperability with legacy VoIP/IMS networks. It also covers ongoing discussions around topics like preferred video codecs and the development of WebRTC browser APIs.
WebRTC Workshop 2013 given at the IMS World ForumAlan Quayle
The document provides an agenda for a WebRTC workshop covering the following key points:
- The workshop will provide a deep technical and business overview of WebRTC through presentations and demonstrations.
- Attendees will learn about the current status of WebRTC standardization and implementations, and what capabilities may emerge over the next 1-2 years.
- The workshop includes sessions on technology details like APIs, media protocols, and interoperability, as well as implications for service providers, enterprises, and use cases.
- An afternoon demo session will provide hands-on experiences of WebRTC applications from various companies and allow networking among attendees.
This document provides an overview and update on WebRTC standards from December 2014. It discusses what WebRTC is, including that it is a browser-embedded media engine. It describes the various WebRTC standards covering signaling, media codecs and protocols. There is no single defined signaling method for WebRTC. The document also discusses topics like the video codec battle between H.264 and VP8, browser support for WebRTC, and interworking WebRTC with legacy VoIP/IMS deployments.
This document provides an overview of WebRTC standardization efforts in 3 sentences or less:
WebRTC standardization is being addressed by the W3C for browser APIs, the IETF for protocols, and other groups like the SIP Forum and 3GPP. There are still open issues around browser support, signaling, interworking with SIP, video codecs, and security. The document concludes by introducing the speaker, Victor Pascual, as a technology consultant who can be contacted for further discussion on WebRTC standardization topics.
Each WebRTC deployment is implementing its own proprietary signaling mechanism. WebRTC signaling and media is incompatible with existing VoIP deployments, requiring gateways to bridge the two. The WebRTC API specification is still evolving and may take different forms across implementations. Standards bodies are working to define signaling interoperability and bridge WebRTC with existing networks like SIP, IMS and videoconferencing systems.
WebRTC provides a standardized profile for real-time communication that enables interoperability between browsers without plugins. It defines client-side APIs for audio and video calling as well as other real-time communication capabilities. The WebRTC architecture includes the API, codecs, transport mechanisms like STUN and TURN, and network I/O that allow real-time apps to run directly in browsers. Signaling is required to establish connections between users, and the standardization of WebRTC aims to improve interoperability compared to proprietary solutions. However, interoperability is not always in the best interests of businesses. Ultimately, the API is more important than the underlying protocols it uses.
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.
The document discusses the Future Internet Assembly (FIA) working group on standardization and pre-standardization. It outlines the working group's action plan to address challenges around standardization gaps, motivate individual contributors beyond single projects, and achieve interoperability. The working group will present standardization activities and roadmaps across several domains including Internet of Things, mobile Internet, testing infrastructure, media/content, trust/security, cloud computing, and services/software.
This document provides an overview and update on WebRTC standards from October 2014. It discusses the key standards bodies working on WebRTC, including the RTCWeb working group. It summarizes the core WebRTC standards for media including supported audio and video codecs, secure RTP, and NAT traversal tools. Open issues are mentioned around the video codec decision and signaling options. Browser support and future directions are also briefly outlined.
In three short webinars, on the morning of April 9, 3GPP presented some conclusions and highlights of the 3GPP Technical Specification Group (TSG) Plenary meetings, held as e-meetings during the week – March 16-20. At 10:00 CET – Lionel Morand, CT TSG Chair, took us through his highlights from the CT#87-E meeting.
An article on this is available on 3GPP website here: https://www.3gpp.org/news-events/2116-ct87_webinar
Video of this is available here: https://vimeo.com/407199676
*** SHARED WITH PERMISSION ***
IMTC Connect 2015, SIP Parity Activity Group UpdateCharles Eckel
1. The SIP Parity Activity Group focuses on enabling migration from H.323 to SIP by providing a video profile for SIP that matches H.323 capabilities and developing best practice documents.
2. A recent interoperability testing event discovered an issue with RTCP feedback interoperability that will require updating the SIP Video Best Practice document.
3. Participation and successful testing results increased in the most recent testing event, though pass rates decreased, and interoperability of newer features like role-based video and security need continued improvement.
WebRTC standards update (July 2014)
The document discusses updates to WebRTC standards in July 2014, including discussions around signaling, video codecs, browser support, and interworking with legacy VoIP/IMS deployments. It notes that each WebRTC deployment implements proprietary signaling, WebRTC signaling and media are not compatible with existing VoIP without gateways, and the WebRTC API could evolve in the future.
The document summarizes a webinar for the 1st Open Call of the symbIoTe project. It introduces the project team members and provides an agenda for the webinar. It then gives an overview of the symbIoTe project including its approach and how IoT platforms can become symbIoTe-enabled. The technical approach for Level 1 compliance is presented, including the components and technologies. Finally, information is provided on how to apply for the 1st Open Call and the application details.
WebRTC will enable real-time communications like voice and video directly in web browsers without plugins. The presenters will discuss their vision for this technology and how to implement it for corporations and telecom networks. They will cover introductions to HTML5, WebRTC, and network architectures; technical challenges around codecs, encryption, and NAT traversal; application cases for telecoms, companies, social media, and manufacturers; and demos of WebRTC applications and identity management. The presentation aims to show how voice traffic will migrate to the web, with browsers as new endpoints and websites as potential call centers, changing how telephone numbers and communications are managed.
NTSC ISDN to IP Video Conferencing Transition RecommendationsVideoguy
The document provides recommendations for educational institutions transitioning from ISDN to IP-based video conferencing on the K-20 Education Network. It outlines key considerations for preparing local networks for IP video, including estimating bandwidth needs, optimizing connectivity, and configuring firewalls and gatekeepers. It also addresses assessing equipment needs, installing and configuring IP video endpoints, and managing video call quality over IP networks. The appendix includes a glossary of terms and checklists to aid in the transition.
This document discusses WebRTC, an effort to build a standard-based real-time media engine into browsers. It examines the status and potential impacts of enabling real-time voice and video capabilities in browsers. WebRTC uses public standards from the IETF and W3C and allows web applications to initiate direct peer-to-peer media connections between browsers or to a media server using HTML and JavaScript. This could transform communications and collaboration over the next five years.
A Webinar by Victor Pascual Avila and Amir Zmora about WebRTC standards. IETF and W3C work on WebRTC as well as interworking with other networks such as IMS. The Webinar also talks about WebRTC signaling options and video codecs.
Status of WebRTC across Asia by Alan Quayle +++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.
Evaluating Wavelet Tranforms for Video Conferencing ApplicationsVideoguy
This report summarizes work done in the second quarter of 2008 on a research project evaluating wavelet transforms for video conferencing applications. The project investigated the DIRAC wavelet-based video codec by [1] developing an understanding of DIRAC's theory and software, [2] creating a software flow diagram for DIRAC, and [3] examining how to integrate DIRAC into an open-source video conferencing application called OpenPhone. The report provides details on DIRAC's wavelet transform, quantization, motion estimation, and entropy coding. It also includes DIRAC's software architecture and classes. Integration of DIRAC into OpenPhone is discussed to test the codec in a video conferencing environment. Future work focuses on
WebRTC standards are being developed by the RTCWeb working group. Key standards include audio and video codecs, secure RTP for media transport, and peer-to-peer connectivity tools. However, WebRTC does not define a signaling protocol, so each deployment implements its own proprietary signaling. While WebRTC media is incompatible with legacy VoIP systems, standards groups are working on interconnection through gateways. The WebRTC API may exist in different versions over time to provide flexibility to web developers.
New IP is a protocol system being developed for Industrial Internet, Industry 4.0, Industrial IoT, etc. This paper clarifies issues raised by an ISOC paper and provides as much information as could be helpful to the community.
When people think about WebRTC, they think about video calls inside a web browser. WebRTC is much more than that. WebRTC can be used to create fundamentally better experiences by embedding live, peer-to-peer communications in SaaS products, mobile apps, and websites. But what is the state of WebRTC today? What does it take for a business to really reap the benefits?
My slide deck from the session I gave at Twilio's Signal event May 2015.
Innovation in the network – Adding value to voice OpenCloud BouyguesAlan Quayle
Innovation in the network – Adding value to voice. TADSummit 12-13 November, Istanbul, Point Hotel Taksim. Patrice Crutel Senior Architect - Core Network & Services, Mark Windle, Head of Marketing, OpenCloud.
This document provides an agenda and overview for a WebRTC workshop. The summary includes:
- The workshop will cover the history, technology, and potential applications of WebRTC, including an overview of the API and standards, demonstrations of real-world services, and a discussion of whether WebRTC is ready for adoption.
- WebRTC allows real-time communication like voice, video, and data sharing directly in web browsers using peer-to-peer connections while abstracting away complexity through the JavaScript API.
- The document discusses topics like ICE, STUN/TURN, security, coding standards, and the ongoing debate around mandatory video codecs.
Thinking the archives of 2020: Opportunitiws, priorities, IssuesFIAT/IFTA
This document summarizes a discussion between members of broadcasting archives organizations about priorities and challenges for archives in 2020. The discussion covered many topics, including storage formats and migration, rights management, metadata automation, user interfaces, and financing models. Participants shared their individual organization's priorities, such as NHK's focus on high resolution content and rich navigation or RAI's projects to digitize archives and automate rights management. Overall, the discussion aimed to identify common issues and opportunities to develop strategies together for the future of broadcasting archives.
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.
The document discusses the Future Internet Assembly (FIA) working group on standardization and pre-standardization. It outlines the working group's action plan to address challenges around standardization gaps, motivate individual contributors beyond single projects, and achieve interoperability. The working group will present standardization activities and roadmaps across several domains including Internet of Things, mobile Internet, testing infrastructure, media/content, trust/security, cloud computing, and services/software.
This document provides an overview and update on WebRTC standards from October 2014. It discusses the key standards bodies working on WebRTC, including the RTCWeb working group. It summarizes the core WebRTC standards for media including supported audio and video codecs, secure RTP, and NAT traversal tools. Open issues are mentioned around the video codec decision and signaling options. Browser support and future directions are also briefly outlined.
In three short webinars, on the morning of April 9, 3GPP presented some conclusions and highlights of the 3GPP Technical Specification Group (TSG) Plenary meetings, held as e-meetings during the week – March 16-20. At 10:00 CET – Lionel Morand, CT TSG Chair, took us through his highlights from the CT#87-E meeting.
An article on this is available on 3GPP website here: https://www.3gpp.org/news-events/2116-ct87_webinar
Video of this is available here: https://vimeo.com/407199676
*** SHARED WITH PERMISSION ***
IMTC Connect 2015, SIP Parity Activity Group UpdateCharles Eckel
1. The SIP Parity Activity Group focuses on enabling migration from H.323 to SIP by providing a video profile for SIP that matches H.323 capabilities and developing best practice documents.
2. A recent interoperability testing event discovered an issue with RTCP feedback interoperability that will require updating the SIP Video Best Practice document.
3. Participation and successful testing results increased in the most recent testing event, though pass rates decreased, and interoperability of newer features like role-based video and security need continued improvement.
WebRTC standards update (July 2014)
The document discusses updates to WebRTC standards in July 2014, including discussions around signaling, video codecs, browser support, and interworking with legacy VoIP/IMS deployments. It notes that each WebRTC deployment implements proprietary signaling, WebRTC signaling and media are not compatible with existing VoIP without gateways, and the WebRTC API could evolve in the future.
The document summarizes a webinar for the 1st Open Call of the symbIoTe project. It introduces the project team members and provides an agenda for the webinar. It then gives an overview of the symbIoTe project including its approach and how IoT platforms can become symbIoTe-enabled. The technical approach for Level 1 compliance is presented, including the components and technologies. Finally, information is provided on how to apply for the 1st Open Call and the application details.
WebRTC will enable real-time communications like voice and video directly in web browsers without plugins. The presenters will discuss their vision for this technology and how to implement it for corporations and telecom networks. They will cover introductions to HTML5, WebRTC, and network architectures; technical challenges around codecs, encryption, and NAT traversal; application cases for telecoms, companies, social media, and manufacturers; and demos of WebRTC applications and identity management. The presentation aims to show how voice traffic will migrate to the web, with browsers as new endpoints and websites as potential call centers, changing how telephone numbers and communications are managed.
NTSC ISDN to IP Video Conferencing Transition RecommendationsVideoguy
The document provides recommendations for educational institutions transitioning from ISDN to IP-based video conferencing on the K-20 Education Network. It outlines key considerations for preparing local networks for IP video, including estimating bandwidth needs, optimizing connectivity, and configuring firewalls and gatekeepers. It also addresses assessing equipment needs, installing and configuring IP video endpoints, and managing video call quality over IP networks. The appendix includes a glossary of terms and checklists to aid in the transition.
This document discusses WebRTC, an effort to build a standard-based real-time media engine into browsers. It examines the status and potential impacts of enabling real-time voice and video capabilities in browsers. WebRTC uses public standards from the IETF and W3C and allows web applications to initiate direct peer-to-peer media connections between browsers or to a media server using HTML and JavaScript. This could transform communications and collaboration over the next five years.
A Webinar by Victor Pascual Avila and Amir Zmora about WebRTC standards. IETF and W3C work on WebRTC as well as interworking with other networks such as IMS. The Webinar also talks about WebRTC signaling options and video codecs.
Status of WebRTC across Asia by Alan Quayle +++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.
Evaluating Wavelet Tranforms for Video Conferencing ApplicationsVideoguy
This report summarizes work done in the second quarter of 2008 on a research project evaluating wavelet transforms for video conferencing applications. The project investigated the DIRAC wavelet-based video codec by [1] developing an understanding of DIRAC's theory and software, [2] creating a software flow diagram for DIRAC, and [3] examining how to integrate DIRAC into an open-source video conferencing application called OpenPhone. The report provides details on DIRAC's wavelet transform, quantization, motion estimation, and entropy coding. It also includes DIRAC's software architecture and classes. Integration of DIRAC into OpenPhone is discussed to test the codec in a video conferencing environment. Future work focuses on
WebRTC standards are being developed by the RTCWeb working group. Key standards include audio and video codecs, secure RTP for media transport, and peer-to-peer connectivity tools. However, WebRTC does not define a signaling protocol, so each deployment implements its own proprietary signaling. While WebRTC media is incompatible with legacy VoIP systems, standards groups are working on interconnection through gateways. The WebRTC API may exist in different versions over time to provide flexibility to web developers.
New IP is a protocol system being developed for Industrial Internet, Industry 4.0, Industrial IoT, etc. This paper clarifies issues raised by an ISOC paper and provides as much information as could be helpful to the community.
When people think about WebRTC, they think about video calls inside a web browser. WebRTC is much more than that. WebRTC can be used to create fundamentally better experiences by embedding live, peer-to-peer communications in SaaS products, mobile apps, and websites. But what is the state of WebRTC today? What does it take for a business to really reap the benefits?
My slide deck from the session I gave at Twilio's Signal event May 2015.
Innovation in the network – Adding value to voice OpenCloud BouyguesAlan Quayle
Innovation in the network – Adding value to voice. TADSummit 12-13 November, Istanbul, Point Hotel Taksim. Patrice Crutel Senior Architect - Core Network & Services, Mark Windle, Head of Marketing, OpenCloud.
This document provides an agenda and overview for a WebRTC workshop. The summary includes:
- The workshop will cover the history, technology, and potential applications of WebRTC, including an overview of the API and standards, demonstrations of real-world services, and a discussion of whether WebRTC is ready for adoption.
- WebRTC allows real-time communication like voice, video, and data sharing directly in web browsers using peer-to-peer connections while abstracting away complexity through the JavaScript API.
- The document discusses topics like ICE, STUN/TURN, security, coding standards, and the ongoing debate around mandatory video codecs.
Thinking the archives of 2020: Opportunitiws, priorities, IssuesFIAT/IFTA
This document summarizes a discussion between members of broadcasting archives organizations about priorities and challenges for archives in 2020. The discussion covered many topics, including storage formats and migration, rights management, metadata automation, user interfaces, and financing models. Participants shared their individual organization's priorities, such as NHK's focus on high resolution content and rich navigation or RAI's projects to digitize archives and automate rights management. Overall, the discussion aimed to identify common issues and opportunities to develop strategies together for the future of broadcasting archives.
Similar to Gn4 1-sa8 t2 tech scout - webrtc2sip-gateway - v1.0-final (20)
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4. SA8T2 Internal Deliverable
Technology Scout: WebRTC2SIP Gateway 3
1 Introduction
1.1 About this document
This document reports on results and findings from a technology scout into a WebRTC-to-SIP
Gateway conducted by the Service Activity 8 (SA8), Task 2 team of the GN4-1 project. This report
should, as such, be read in context of the related work produced by GN4-1 SA8-T2.
The WebRTC task ran from 1 May 2015 to 30 April 2016.
1.1.1 Target audience
This document targets technical management and specialists, in particular those working in the
fields of real time communications, eLearning and eResearch.
1.1.2 Responsible task members
Stefan Otto (UNINETT) had the lead on this tech scout. Jan Meijer (UNINETT) and Simon Skrødal
(UNINETT) were the document editors.
1.2 Background
The Session Initiation Protocol (SIP) is a system of rules that governs the signalling and controlling of
multimedia communication sessions. This technology scout investigates how the WebRTC ecosystem
may integrate and support legacy SIP equipment.
There are a number of scenarios where it would make sense to enable audio/video communication
between legacy SIP clients and web browsers. While participation in a classical SIP video meeting
with a WebRTC-enabled browser is already possible, the other way around is not. You can not
participate in a WebRTC browser meeting using a SIP client. This will never work via a universal
gateway because there is no standardized signalling protocol in WebRTC, meaning that you would
need to implement your own gateway for each WebRTC-service.
Other, more feasible and value-adding use cases include:
5. Introduction
SA8T2 Internal Deliverable
Technology Scout: WebRTC2SIP Gateway 4
• easy to integrate call buttons where web page visitors can ring to your organization's
internal telephony network by pressing a button in their browser
• participate with a browser in a classic audio only conference
• build Point-of-Presence systems which can interact both with classical telephony and legacy
video systems
• wherever you want to connect a browser to telephony or legacy SIP video systems
WebRTC and SIP media sessions have many similarities (see Figure 1: WebRTC Signalling and media
and Figure 2: SIP Signalling and media). Since WebRTC standardization began, there have been
ongoing discussions and efforts towards maintaining compatibility with classic SIP audio/video
equipment. For this reason, there are no specific signalling protocols defined for WebRTC, meaning
you can use SIP if you want. Further, the Session Description Protocol (SDP) is used by SIP as well as
WebRTC to describe media session and communication parameters, such as key-fingerprints and IP-
address candidates. In WebRTC, this SDP "blob" has to be sent with the signalling protocol of your
choice. Hence, there are several similarities worth investigating further in order to bridge WebRTC
and SIP.
Figure 1: WebRTC Signalling and media
Figure 2: SIP Signalling and media
7. SA8T2 Internal Deliverable
Technology Scout: WebRTC2SIP Gateway 6
2 Challenges
A closer look identifies a number of issues that must be addressed.
2.1 Transfer Protocol
SIP-clients use direct TCP or UDP to transport SIP packets. The web browser, on the other hand, can
only communicate using HTTP/WebSocket. Since the web developer cannot make the web browser
send or receive SIP-messages to/from a specified IP-address/port, the transfer protocol must be
converted. This may be achieved by using the WebSocket protocol as transport for SIP (see
https://tools.ietf.org/html/rfc7118) or to translate to SIP on the WebRTC-signalling server and send
forwards.
2.2 Codecs
The next challenge involves the video/audio codec negotiation. Current WebRTC-enabled browsers
(e.g. Firefox/Chrome) support the following codecs:
Video — VP8 (MTI), VP9 and H264 (MTI - supported on Firefox, experimental support in Chrome-
M50)
Audio — opus (MTI), G.711 (MTI), iSAC and G.722
SIP clients usually understand several codecs, but if there is no single match to the ones from the
browser they will not be able to talk to each other. A gateway could decode and encode the streams
to the supported codec, but this would add significant CPU workload, additional latency and
compatibility issues.
2.3 Encryption
For WebRTC connections it is mandatory to use DTLS to establish a secure SRTP connection. Only a
few legacy systems support this.
10. Possible candidates
SA8T2 Internal Deliverable
Technology Scout: WebRTC2SIP Gateway 9
3.2 Doubango webrtc2sip
Doubango Telecom is a young telecom company located in France, with a big focus on open source
solutions in IMS/SIP and WebRTC area. SIPml5, a browser JavaScript SIP library, is provided by
Doubango Telecom.
Their Doubango webrtc2sip media gateway (https://www.doubango.org/webrtc2sip/) consists of 4
modules (SIP Proxy, RTCWeb Breaker, Media Coder, Click-to-Call) following the same philosophy as
described in this document. Additionally, it features audio/video transcoding abilities. In our tests
one year ago we experienced the solution as slightly unstable. There are no binary packages, so you
have to build everything on your own. After updating openSSL due to security issues, we were no
longer able to build it and thus ended our testing of webrtc2sip at that point.
3.3 Janus
The Italian company Meetecho, located in Naples, provides an open source general purpose
WebRTC Gateway called Janus (https://janus.conf.meetecho.com). Janus consists of a small
footprint core, which can be extended/customized with a number of plugins. There are already a few
plugins available, e.g. audiobridge, echotest, recordplay, SIP, streaming, videocall, videoroom and
voicemail. The SIP plugin enables peers to register through Janus to a SIP server and call in and out.
The functionality is marked as unstable and need further testing - for now tested successfully audio
only.
The SIP plugin is based on Sofia SIP (http://sofia-sip.sourceforge.net). No documentation was found
in regard to SDP translating between legacy SIP equipment and WebRTC.
3.4 Kamailio + rtpengine
The author of this document is part of the Norwegian real-time group at UNINETT, and we built a SIP
infrastructure based on the solid SIP router Kamailio (https://www.kamailio.org). It therefore made
sense to focus our attention on a SIP-proxy solution based on Kamailio.
11. Solution Overview
SA8T2 Internal Deliverable
Technology Scout: WebRTC2SIP Gateway 10
4 Solution Overview
Figure 4: WebRTC2SIP-Gateway overview
The implemented solution utilizes Kamailio as the SIP-proxy and Sipwise rtpengine
(https://github.com/sipwise/rtpengine) as the RTP-proxy. Kamailio controls rtpengine with the
integrated rtpengine module to facilitate the following: Kamailio passes a received SDP body to
rtpengine à rtpengine rewrites the body and returns it à Kamailio forwards the rewritten body.
This flow guarantees that each SIP-client (WebRTC SIP.js clients as well) gets only ICE candidates
from rtpengine. In result RTP traffic will flow via rtpengine. Additionally, rtpengine takes care of
BUNDLEd media and muxed RTP/RTCP channels and rewrites these lines in the SDP body as well.
Hence, rtpengine effectively talks to each client side and translates in the middle (this does not
include media transcoding).
The web server is the entry point for the web browser and its naming should be user friendly (e.g.
call-the-ancient-world.net). Running the HTML/Javascript-code from the web server turns the
browser into a WebRTC SIP client that registers on Kamailio. The client is then callable via a SIP-URI
world-wide or can place its own calls. In this example configuration, the SIP-URI is websip@gateway-
domain.no
Should the WebRTC client be unable to connect directly to rtpengine it will receive support from the
TURN-server, which listens on TCP/TLS port 443 and forwards the media traffic to rtpengine via UDP.
This is needed to pass restrictive firewalls on the client side. rtpengine does not support RTP media
over TCP for now — if this will be the case in the (near) future, there will no longer be a need for a
TURN server.
15. SA8T2 Internal Deliverable
Technology Scout: WebRTC2SIP Gateway 14
7 Security considerations
User credentials for the SIP-proxy and for the TURN server are publicly available from the web
server. Some kind of authentication should therefore be established in front of the web server. The
TURN server should be protected with time-limited secret-based authentication (see A REST API For
Access To TURN Services, https://tools.ietf.org/html/draft-uberti-behave-turn-rest-00) or an oAuth-
based client authorization, which is experimentally supported by the coTURN TURN server. To
protect Kamailio, we could generate user accounts dynamically on the web server and push these to
Kamailio’s user-database.
Since media traffic is decrypted in the Gateway, it is possible to intercept the media traffic between
the gateway and legacy SIP equipment. Metadata interception is possible if you intercept network
traffic to and from the gateway. You can get IP-address based call information (when, how long, SIP
header information).
18. Conclusions
SA8T2 Internal Deliverable
Technology Scout: WebRTC2SIP Gateway 17
git clone https://github.com/havfo/WebRTC-to-SIP.git
cd WebRTC-to-SIP
find . -type f -print0 | xargs -0 sed -i 's/XXXXX-XXXXX/PUT-IP-OF-YOUR-SIP-
SERVER-HERE/g'
find . -type f -print0 | xargs -0 sed -i 's/XXXX-XXXX/PUT-DOMAIN-OF-YOUR-
SIP-SERVER-HERE/g'
find . -type f -print0 | xargs -0 sed -i 's/XXX-XXX/PUT-DOMAIN-OF-YOUR-
TURN-SERVER-HERE/g'
A.3 Install rtpengine
The following achieves the SRTP-RTP bridging needed to make WebRTC clients talk to legacy SIP
server/clients:
apt-get install dpkg-dev debhelper iptables-dev
libcurl4-openssl-dev libglib2.0-dev libhiredis-dev
libpcre3-dev libssl-dev markdown zlib1g-dev
libxmlrpc-core-c3-dev dkms linux-headers-`uname -r`
git clone https://github.com/sipwise/rtpengine.git
cd rtpengine
./debian/flavors/no_ngcp
dpkg-buildpackage
cd ..
dpkg -i 'ngcp-rtpengine-daemon_*' 'ngcp-rtpengine-iptables_*' 'ngcp-
rtpengine-kernel-dkms_*'
cd WebRTC-to-SIP
cp etc/default/ngcp-rtpengine-daemon /etc/default/
/etc/init.d/ngcp-rtpengine-daemon restart
A.4 Install IPTables firewall (optional)
Optional, but recommended to secure your servers. It will work on all three servers, but you should
customize _iptables.sh_ for each of them (e.g. the web server does not need the TURN, RTPENGINE
and RTP rules, while the TURN server does not need the RTPENGINE rule). If installed it will persist
after reboot. You can run the _iptables.sh_ at any time after it is set up.
cd WebRTC-to-SIP
chmod +x iptables.sh
cp etc/network/if-up.d/iptables /etc/network/if-up.d/
chmod +x /etc/network/if-up.d/iptables
touch /etc/iptables/firewall.conf
touch /etc/iptables/firewall6.conf
./iptables.sh
A.5 Install Kamailio
apt-get install kamailio kamailio-websocket-modules kamailio-mysql-modules
kamailio-tls-modules kamailio-presence-modules mysql-server
cd WebRTC-to-SIP
19. Conclusions
SA8T2 Internal Deliverable
Technology Scout: WebRTC2SIP Gateway 18
cp etc/kamailio/* /etc/kamailio/
kamdbctl create
Select yes (Y) for all options.
kamctl add websip websip
/etc/init.d/kamailio restart
A.6 Install WebRTC client
apt-get install nginx
cd WebRTC-to-SIP
cp etc/nginx/sites-available/default /etc/nginx/sites-available/
cp -r client/* /var/www/html/
A.7 Install TURN server
apt-get install rfc5766-turn-server
cp etc/default/rfc5766-turn-server /etc/default/
cp etc/turn* /etc/
/etc/init.d/rfc5766-turn-server restart
A.8 Testing
You should now be able to go to https://webrtcnginxserver/ and call to legacy SIP clients.