The Week that was in WebRTC: Gateways

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By Tsahi Levent-Levi

May 3, 2013  

What a week… just when I complain about gateways – they get announced en mass.

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I recently published my first post in UC Strategies. Being a unified communication domain, I decided to complain there about the fact that a WebRTC gateway strategy is nice but not enough.

As if that wasn’t a reminder enough, we’ve been bombarded by 3 press releases about WebRTC gateways this week:

That’s a nice list of industry first/leading/comprehensive, but it begs the question – how can that be when there are some other WebRTC gateways out there?

The ones I am aware of are:

  • Doubango Telecom, offering a WebRTC to SIP gateway, and an open source one while we’re at it
  • Mavenir, who announced their gateway a week or two ahead of Huawei’s “industry first”, for IMS
  • Thrupoint, who have shown their WebRTC to SIP gateway last year already
  • And then there’s Acme Packet/Oracle, with their SBC (which is a kind of a gateway) that supports… WebRTC and IMS for half a year now

Know any other gateway vendor with WebRTC that I’ve missed somehow?

Anyone else who wants to say how hard it is to interoperate with SIP when we already have 7 different gateways out there?

Oh… and yes… WebRTC infrastructure is tricky (my contribution this week to NoJitter), but a gateway isn’t what I have in mind as the solution to it.

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  4. Tzahi, read your post in UC Strategies. Laughed much. It’s written in a very sharp and provocative style. Which is good as it was aimed to serve a “Wake-up call”. The comments from the community are also good. I wish you could answer one gentleman there who tried to figure out how you would maintain call set-up, call control and call routing. How WebRTC treats network topology, routing, authentication, etc?

    Reply

    1. David,

      I plan on touching the way I think UC vendors should treat WebRTC in my next post on UC Strategies.

      As for the “boring” issues of network topology and routing – these are out of scope of WebRTC, which brings a lot of power to vendors to differentiate and get creative. As for authentication – why not go for whatever is used in the enterprise the offering is used for? Be it Google profile if it is a Google Apps enterprise or Active Directory for a Microsoft shop.

      Reply

  5. Tzahi, read your post in UC Strategies. Laughed much. It’s written in a very sharp and provocative style. Which is good as it was aimed to serve a “Wake-up call”. The comments from the community are also good. I wish you could answer one gentleman there who tried to figure out how you would maintain call set-up, call control and call routing. How WebRTC treats network topology, routing, authentication, etc?

    Reply

  8. Phono http://phono.com provides a hosted or on-premise gateway for signaling, media, and interop with SIP, pstn, and Jingle, and a set of media APIs for non-WebRTC endpoints via Flash, mobile, and Java clients. Also a simple jQuery plugin for building apps on these media APIs and WebRTC. The Phono jQuery API has even been adopted by Ericsson and AT&T for their WebRTC products.

    Reply

  9. Hi Tsahi,
    just stumbled upon this article. I wonder, almost one year and a half later, what’s your current view, and whether you have an up to date list of gateway providers.
    Thanks,
    Giacomo

    Reply

    1. Giacomo,

      I don’t have any updated list of gateway providers, at least not an easy to consume one.

      Out of the top of my head, if you want to commnuicate with SIP, one of the options is simply to use FreeSWITCH or Asterisk – they both offer WebRTC support with SIP over WebSocket signaling. There’s also Voxbone, who provide dialing services from WebRTC to PSTN numbers (assuming they are yours). And there’s the generic Janus gateway which enables you to “fill it in” with your own implementation.

      Also check at http://webrtchacks.com/vendor-directory/

      Tsahi

      Reply

      1. Thanks Tsahi,
        I’m familiar with Asterisk and FreeSWITCH (and Kamailio, which does a good job with WebSockets) as I come from a “traditional” VoIP background. I had also read about Voxbone’s offering in this field.
        I’ll look up the Janus gateway (and the list of vendors, which I’ve just skimmed now).
        An aspect around WebRTC and PSTN that I’m interested about at the moment – and wasn’t stated clearly in my previous comment – is how to get from browser to PSTN(/GSM) without the client “talking SIP”.
        I appreciate it’s a wide topic and probably deserves a separate discussion.

        Thanks again for your reply,
        Giacomo

        Reply

        1. Giacomo,

          Out of the top of my head, that would be done by doing the interoperability/translation part in the backend using a … gateway.

          Twilio, for example, uses proprietary signaling towards the client but has a SIP backend.

          In such a case, the best approach is probably to use something like Janus and build whatever it is you are looking for into it. Read this one as well: http://webrtchacks.com/webrtc-gw/

          Reply
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