Steve Johnson and Karl Stahl of Ingate invited me to moderate a massive panel at their SIP Trunking, UC & WebRTC Seminars at ITEXPO last week, and I have to say that I was ill prepared for what happened at the event.
Steve Johnson and I had talked about the need to get some perspective on WebRTC, and what it does and does not do. Me, I am a big advocate for WebRTC because it expands the role of communications on the web. I can see it impacting our business opportunities in so many ways. I don’t, though, see it as a replacement for the PSTN, and from that standpoint I don’t even see it replacing SIP. (Truth be told, in fact, I think it would be upsetting to see it replace Skype! Not because it can’t, but because it shouldn’t.) What I found weird, though, was that a fellow WebRTC advocate was ribbing the SIPsters about the capacity and capability of WebRTC in comparison to SIP, and that some of the SIPsters took the bait and started dismissing the value of WebRTC.
Now I am a moderation-in-all-things kind of guy, save of course for Diet Coke and garlic bread, but this sounded like Kierkegaard yelling out “Either / Or”.
SIP has suffered greatly in its efforts to make the IETF the place where telephone standards are managed. If it was not for SIP, in fact, the possibility of turning off the circuit switch network would never have happened. In taking on this burden, however, the bloat of the standard is obvious to all, though it makes the standard interoperable, universal and compliant with regulatory requirements. WebRTC on the other hand is light, innovative and currently exempt from all the compliance issues faced in telecom. Thus, it is obvious that WebRTC looks like fun while SIP looks like…well, POTS.
Where we should be focusing on is what WebRTC does and how SIP can take advantage of it. In my opinion, WebRTC can be looked at as a gateway for web developers to understand how to enable communication. It can add all sorts of wonderful enhancements, all of which I have cited in this column several times, and all of which are driven by WebRTC’s APIs and deliberate design as a Web tool.
- Speech Recognition
- Web(Text) to Speech
- Advertising “Call” Sessions on remote sites
- Interactive Community Communication
As a web tool, WebRTC is universal and ubiquitous because, in effect, it relies on the browser to connect the end points. It is, however, not interoperable with the PSTN, SIP, or anything we would call telecommunications. Now, to get to the point where a third party can be reached you need to come back to SIP. And because SIP handles signaling, it often gets compared to Node.JS, MQTT, Websockets, and XMPP, though only XMPP is in the ballpark of trying deliver outside of a domain. The other signaling protocols are formidable, delivering scalability, but they do not answer the question, “What if I wanted to reach beyond the domain of the website?”
Now SIP can gain a lot from WebRTC making communications fun for web developers and not having to explain ice, stun, turn and trapezoids. As such, the debate should not be about “Either / Or” but about what can be done to get developers working to make the web and telecommunications play well together.
Edited by Stefania Viscusi