How Will WebRTC Change the VoIP Industry?

This is another one of those posts about WebRTC. The ones you might be interested in?

• What is WebRTC?
• How will WebRTC Change the Video Conferencing Industry?

When I published my post on what WebRTC will do to Video Conferencing, a colleague of mine asked what will it to the larger VoIP industry.

I think that this question renders a similar but a bit different answer in such a case. The focus here changes a bit – we’re not interested in video – only in the voice capabilities of WebRTC. And voice has become a commodity already.

I’d like to highlight 3 ways in which the VoIP industry is being changed by WebRTC:

Shift in the power structure of voice engine offerings in the market

From the moment Google has acquired GIPS (the company that was later open sourced as WebRTC), their user base has scrambled to search for alternatives. GIPS was one of the most known and successful voice engine companies – almost every company developing a VoIP product used them.

It was somewhat expected that Google would open source them, but that has left the issue of an SLA wide open – there was no real way you as a customer can get an SLA from Google for maintaining and improving the GIPS voice engine any longer. And it is an issue, as a lot of GIPS customers used customized voice engine packages – ones that fit specific hardware and embedded processors.

With a limited number of alternatives around, companies had to find other solutions and at times invent and invest in their own home grown solutions. No fun there.

This shift of power can be beneficial to some:

• Companies who were lacking a good voice engine solution but were willing to invest engineering resources on it now have a great starting point – they can take WebRTC and embed it into their own products – web based or not.
• Small software vendors can now offer engineering services around WebRTC or rather GIPS’ earlier solutions, providing an SLA for those searching for it.

Addition of new voice transcoders

If your product deals with voice codecs, it should probably now add those supported by WebRTC inherently. These codecs are different than the ones used in most SIP deployments. While there is a common denominator – G.711 – that’s a poor man’s solution.

This means that VoIP products now need to add iSAC and maybe iLBC to their arsenal. Not hard, but still a pain.

“Translators“ from SIP to… web-something

As WebRTC has no real signaling attached to it, it is half the solution. The rest of the VoIP world is still mostly SIP (I am ignoring Skype and Gtalk on purpose here), so to offer something here, companies will need a kind of a gateway from SIP to the open ended JavaScript signaling of WebRTC.

Why is that important? Once solutions that use only WebRTC start surfacing, there will be a need to connect them to the “other” world. That of SIP and PSTN. And this requires gateways.

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It is going to be interesting to see how this evolves moving forward.