On the Road Again – SIPNOC 2014

I’ll be speaking next week at the SIPNOC conference in Herndon, Virginia.  SIPNOC is sponsored by the SIP Forum and covers a wide variety of topics related to SIP — the Session Initiation Protocol — with a particular focus on the needs of service providers.   It runs from June 9 – 12.

WebRTC continues to be a hot topic in the telecom industry and I’ll be on a panel with several other participants to discuss the relationship between SIP and WebRTC.   SIP has been the primary protocol for Voice over IP and is widely deployed.  WebRTC is much newer, but offers an interesting mix of audio, video and data capabilities and it can be accessed via popular browsers from Google and Mozilla.  WebRTC also has a rapidly growing eco-system.  Are SIP and WebRTC complementary standards which work well together or going in totally different directions?  Come to the panel and find out!

I am also delivering a presentation on a very exciting development in IP fax communications over SIP.  The presentation is entitled: Securing IP Fax – A New Standard Approach.  It’s been a long time coming, but there will soon be a new security standard for implementors of IP Fax over SIP networks.  In particular, the Internet Engineering Task Force is working on using an existing security standard known as DTLS and adding this as a security layer for T.38 fax.    I’ll be talking about the pending standard, why it’s needed and what kind of benefits can be expected for the many users of T.38 IP fax once the new standard is deployed.

I’ve attended SIPNOC as a speaker since its beginning four years ago.  It’s an excellent conference and offers an in-depth perspective on the latest news in SIP as delivered by an all star cast of speakers.  I hope you’ll be able to join us.

WebRTC – Solution for Over The Top Communications?

WebRTC offers an intriguing mix of web-based access and real-time communications.   Part of the excitement has been due to the aggressive approach which has been taken by browser companies such as Google and Mozilla in adding WebRTC to recent versions of their production browsers. 

As a result, any user of these browsers could potentially be connected to other users of WebRTC applications. One example where this could come into play is in Over the Top (OTT) applications. The term Over the Top usually means that an application runs over a broadband IP network and is usually not a packaged service sold by the Internet service provider (ISP). For example, Skype provides a way to do audio and video communications over IP networks. Its base level of service allows for connection to other Skype users at no charge for both audio and video communications. Skype also includes sophisticated features like encryption of calls. For ISPs, Skype potentially competes with a bundled voice offering and a user might elect to use the combination of Skype and a mobile phone for all of their voice communications. This means the ISP gets to sell the customer a broadband IP connection, but may not get any other bundled service revenue.

Let’s suppose you’re an ISP that would like to offer an alternative to Skype for your customer community. What does WebRTC bring to the table? On the media side, WebRTC can support both audio and video communications. It also has built-in security methods for authentication and securing of sessions. For the application, the ISP can create this from scratch or layer this onto a WebRTC enabled browser and automatically take advantage of the WebRTC “hooks” which are built into a browser such as Chrome or Firefox. To truly complete the OTT application, there is still more to do such as determine which signaling to use, and what addressing scheme should be used to interconnect users. For a good analysis of the signaling side, see this recent blog post from webrtchacks.

So, let’s assume the ISP completes the OTT application using WebRTC. What is the potential value add compared to a application like Skype? One potential benefit is the capability for the user to communicate with other users that have WebRTC-enabled applications. One limitation of Skype is that it is a closed community and uses proprietary technologies. As a result, Skype users can currently only communicate with other Skype users unless they go off the network. By contrast, with WebRTC, there will be a standards-based interface based on JavaScript APIs, so that the ISP could structure their application so that it can talk to other WebRTC-enabled applications. There are also a wide variety of WebRTC to SIP gateways that have already been brought to the market, so this offers the potential to interconnect the WebRTC enabled application with the existing base of SIP applications. Hence, WebRTC offers the potential to help break down the silos which currently dominate multimedia communications and enable different applications to communicate either directly via WebRTC or indirectly through WebRTC to SIP gateways.

One way to look at WebRTC is that it offers a very robust “toolkit” of multimedia communications capabilities that can run over web interfaces. The example we have discussed in this blog of an OTT application is just one possibility of how a developer or ISP might use this toolkit. As the web development community learns to take advantage of WebRTC, there will no doubt be a wide range of applications which will emerge. On the business side, WebRTC is a disruptive technology, so we can also anticipate a wide array of different business models to emerge which will build on its open standards hooks.