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troubleshooting (Image source: google)

troubleshooting (Image source: google)

WebRTC-based services are seeing new and larger deployments every week. One of the challenges I’m personally facing is troubleshooting as many different problems might occur (network, device, components…) and it’s not always easy to get useful diagnostic data from users.

Earlier this week, Tsahi, Chad and I participated at the WebRTC Global Summit in London and had the chance to catch up with some friends from Google, who publicly announced the launch of test.webrtc.org. This is great diagnostic tool but, to me, the best thing is that it can be easily integrated into your own applications; in fact, we are already integrating this in some of our WebRTC apps.

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Last year we interviewed Oleg Moskalenko and presented the rfc5766-turn-server project, which is a free open source and extremely popular implementation of TURN and STURN server. A few months later we even discovered Amazon is using this project to power its Mayday service. Since then, a number of features beyond the original RFC 5766 have been defined at the IETF and a new open-source project was born: the coTURN project.

Today we are catching up  with Oleg again to see what’s new and to learn what coTURN is about.

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As discussed in previous posts, WebRTC standards do not specify a signaling protocol. In general this decision is positive by giving developers the freedom to select (or invent) the protocol that best suits the particular WebRTC application’s needs. This can also reduce the time to market since standards compliance-related tasks are minimized. WebRTC media and data protocols from the provider to the user are standardized, so the lack of a standardized signaling protocol does not hurt interoperability of subscribers within the same network service. The calling party just has to have a URL from the called party to download its web app and to establish a WebRTC session with them. They both connect to the same web server and will then utilize the same signaling scheme. This is a new paradigm that is often difficult to embrace for traditional telephony developers who are used using the SIP protocols for handling all signaling, including the User to Network Interface (UNI) and Network-to-Network Interface (NNI).

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One of the first posts we published on this blog a year ago was a ‘A Hitchhiker’s Guide to WebRTC standardization‘. Since then, the work has certainly progressed and we have been sharing here a number of updates on the topic. This week we’re having qn IETF meeting in Canada and when it comes to WebRTC some of the topics in the agenda include ALPN (Application Layer Protocol Negotiation)STUN Consent Freshness and audio (interop with legacy) and video (H.264 vs VP8 as mandatory codec is NOT discussed this week but you can see VP9 and H.265 are already mentioned in the slides) requirements. Several Working Groups other than RTCWeb will also discuss this week topics that are relevant to the WebRTC effort (e.g. MMUSIC WG). During the STRAW WG session, the working group I co-chair at the IETF, we’ll discuss some features of interest for those implementing WebRTC in servers like MCUs, Application Servers, WebRTC-SIP gateways or WebRTC-enabled SBCs: DTLS-SRTP, STUN and RTCP traversal/termination are examples.

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We probably talk about existing telephony stuff too much here, but the reality is there are billions of phone about there that want to be connected to the web like nearly everything else. This is especially important for any business that wants to link its website with its internal phone system.

If you are a modern enterprise or contact center, odds are you connect your phone system to the PSTN using a SIP trunk. This approach is light years ahead of previous technologies, but what about allowing callers to come in from the web with WebRTC? Often this means buying your own gateway or connecting your website to a hosted API provider. Is there something like a WebRTC trunk?

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photo credit Flicr user mattallen89 (CC license)

photo by Flickr user mattallen89 (CC BY)

WebRTC promises to greatly simplify the development of multimedia realtime communications, without the need to install an application or browser plug-in. It enables this by exposing a media engine and the network stack through a set of specialised APIs. Application developers can use these APIs to easily add realtime communication to web applications. The defined use cases that have been identified in the early stages of the standardisation have a wide range of applicability. Many of the initial applications focus on communications between applications running on web browsers, but it is expected that WebRTC will also allow for interoperability with several existing communications protocols. For that and other reasons, the signalling interfaces have been determined to be out of scope, thus allowing the service providers to choose the best suited protocol or signalling mechanism for their users.

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The first post we published on webrtcHacks was ‘A Hitchhiker’s Guide to WebRTC standardization’ (July 2013) where we gave some initial insight on activities in the 3GPP around WebRTC and  IMS. Since then the situation has certainly evolved (well, probably not as fast as some would have expected). Since we regularly receive emails asking about the status/progress on WebRTC standardization within the 3GPP, we spent some time with our friend Antón Román, CTO at Quobis and author of the popular post ‘Anatomy of a WebRTC SDP’ to summarize the current status of the ‘WebRTC access to IMS’ effort.

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As I mentioned in my ‘WebRTC meets telecom’ article a couple of weeks ago, at Quobis we’re currently involved in 30+ WebRTC field trials/POCs which involve in one way or another a telco network. In most cases service providers are trying to provide WebRTC-based access to their existing/legacy infrastructure and services (fortunately, in some cases it’s not limited to do only that). To achieve all this, one of the pieces they need to deploy is a WebRTC Gateway. But, what is a WebRTC Gateway anyway? A year ago I had the chance to provide a first answer during the Kamailio World Conference 2013 (see my presentation WebRTC and VoIP: bridging the gap) but, since Lorenzo Miniero has recently released an open source, modular and general purpose WebRTC gateway called Janus, I thought it would be great to get him to share his experience here.

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Next week Mobile World Congress (MWC) 2014 will take place in Barcelona, Spain. Since Barcelona is my hometown, it’s always a great opportunity to meet with industry friends and enjoy some local spots together.

Many webrtcHackers will be in Barcelona for the event, so we are organizing a meetup next Tuesday at 6PM CET. This event will be largely a social meet & greet but we will have some structured conversation to discuss the latest in WebRTC and developer needs. We will update with exact location details within the Fira Gran Via prior to the event but we will likely meet in one of the food court areas between center the exhibit halls.

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Gustavo Garcia Bernardo

Gustavo Garcia Bernardo

WebRTC and its peer-to-peer capabilities are great for one-to-one communications. However, when I discuss with customers use cases and services that go beyond one-to-one, namely one-to-many or many-to-many, the question arises: “OK, but what architecture shall I use for this?”. Some service providers want to reuse the multicast support they have in their networks (we are having fun doing some experiments with this), some are exploring simulcast-based solutions, others are considering centralised solutions like MCUs/mixers, and a bunch of them are simply willing to place the burden on the endpoint by using some variation of a mesh-based topology.   The folks at TokBox (a Telefónica Digital company) have great experience with multiparty conferencing solutions.  I thought it would be great to have my friend Gustavo Garcia Bernardo (Cloud Architect at TokBox) to share here his take on the topic.

At TokBox, Gustavo is responsible for architecture, design, and development of cloud components. This includes Mantis, the cloud-scaling infrastructure for the OpenTok, which uses the WebRTC platform. Before joining TokBox, Gustavo spent more than 10 years building VoIP products at Telefónica and driving early adoption of WebRTC in telco products. In fact, I’ve known Gustavo for 8 years now and the first time I met him it was preparing a proposal for a European Commission-funded research project on P2PSIP. Since then we’ve been collaborating in the IETF doing some work in the context of P2PSIP, ALTO and SIP related activities. A couple of years ago, while I was working with Acme Packet (now Oracle), we worked together designing and launching Telefonica’s Digital TuMe and TuGo.  Lately we have both shifted our focus towards WebRTC.

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