The WebRTC part of the SIP Survey
For us, those that deal with WebRTC daily, it sometimes seems as if everyone knows all about WebRTC and its promise. It is when you get to read opinions of people that deal with existing telephony systems that you understand this it is not the case.
This post is about one such case.
The SIP Survey
At the end of 2015 The SIP School released their yearly SIP Survey. 1098 responders, many of them enterprise IT professionals and some ITSPs.
Most of the survey was around SIP Trunks but the last part of it was about WebRTC.
On the SIP side of things you can see that many of the enterprises are still using “traditional” on-premise PBXs such as Avaya and Cisco. This kind of sets the tone for the answers with regards to WebRTC.
The first question related to WebRTC was about how well are people familiar with WebRTC.
The shocking number is not that almost 70% are only familiar with it in conceptual basis because these 70% already know what it is and might have even used it here or there.
They know enough about it to know that they need to learn more.
The shocking number is 19.86% that hear about it for the first time.
The following 3 questions talk more about the deployment and adoption of webRTC. I touched this point together with a few WebRTC pundits with regards to ORTC. The questions in this survey were more general.
Here are the results and my opinion on these questions.
Will WebRTC be hit by the Bell Heads?
The survey question was:
Do you think WebRTC will be ‘allowed’ to flourish as a Peer to Peer technology or do you think it will follow the same path as SIP and require multiple intermediary devices such as Session Border Controllers to work?
Peer to peer
In reality WebRTC is Peer to Peer to the exact same extent SIP is. It even uses the same technology as SIP for NAT traversal (ICE).
The reason why we don’t see too many peer to peer SIP deployments is not because of the protocol but because of those who use it to build products and services. SIP is normally deployed in kind of walled gardens where each vendor has his own flavor and therefore there is a need for some mediation on the signaling side and sometimes also on the media side (in many cases for control reasons). Hence, in most cases communication is not peer to peer.
Islands of communication
Most of the respondents (46.54%) said they don’t know while 31.44% said NO.
Reality is that WebRTC was built to be integrated into web services. It wasn’t built for telcos. The absence of signaling makes this point even stronger. It means that as long as WebRTC is being used as part of a web service (e.g. a multi-party game or an eCommerce site) there is no need for mediation regardless what the Bell Heads “allow” or don’t “allow”.
WebRTC moves communication to the hands of the Net Heads.
In the cases where it is added as an interface to existing telco services, mediation will be required but frankly speaking, it is not that much interesting. Mediation is used in those networks anyway.
The survey question was:
What do you see are the biggest challenges with using WebRTC?
I find the answers to this question pretty strange.
I think that the answers represent a telco perspective and that is why they miss the point.
Security is one of the advantages of WebRTC. I don’t know of any other VoIP technology that is being deployed in a more secured way.
While encryption of media and signaling is possible in SIP, Most deployments are still not encrypted.
In WebRTC there is no other option. Media must be encrypted and the road is leading us to encryption of signaling as now getUserMedia in Chrome works only with websites that use https.
The interoperability thing is a question of perspective. There is no need for interoperability in WebRTC, only API and media compatibility so all browsers will be able to run the web app.
As mentioned above, if we are talking about a Web application scenario, interoperability is not a requirement.
In the traditional telco type of scenarios there is that intermediary device as there is a need to GW the traffic anyway.
Is WebRTC a risk to service providers?
The survey question was:
This question is designed to get your comments and thoughts so we can see what you really think WebRTC can do, here we go: Do you think WebRTC services will pose a threat to SIP trunking in the future?
The answer is yes simply because WebRTC moves traffic from those SIP trunks and service providers’ networks to the internet. It becomes part of the web world and telcos are not part of it.
Some telcos are making the right steps by creating the right environments for developers to build web communication services using their networks or creating asymmetric business models that will allow them to make money not only from per minute type of service models.
Why this is important
Many of those coming from IT or traditional communication companies view WebRTC as yet another VoIP protocol.
This is not the case. It is a technology for web developers to use in web applications.
Once this resonates, understanding the risk and potential are easier.