← Back to context

Comment by paulej

11 years ago

I was the editor of the H.323 spec years ago. Funny you would say that SIP looks like it comes from the telcos, when the whole aim with SIP was to create just the opposite. H.323 has often been accused of being created by telcos, too. Truth is, folks from the traditional enterprise voice equipment makers shaped both.

SIP was adopted by 3GPP in 1999 to be used in the mobile networks. Of course, that trickled throughout the telcos, so it now is widely used in the telco space. H.323 was widely used for PSTN backhauling to reduce tolls. However, SIP is more prevalent there, I'd guess.

Enterprise networks still use both H.323 and SIP, with more desk phones using SIP and videoconferencing equipment split, but moving to SIP.

That said, any move to SIP these days is only because there is nothing better. WebRTC is the next big thing. It borrows SDP from SIP, but no signaling protocol is defined.

Personally, I'd like to realize the vision behind what was intended to be H.325, which was to be a JSON-based signalling protocol that is fully distributed. Each application could implement whatever protocol it wanted, and they would all be associated through H.325, giving the user the feeling that these different applications work closely together. That would be possible, even when applications run on different devices.

Alas, there seems to be no financial motivator for those that control the market to explore that concept, so we're stuck with H.323, SIP, and WebRTC (and various proprietary glue pieces). The trend these days seems to be toward proprietary solutions, providing limited interop with others through SIP. Limited, as in nothing but basic voice and video.

I recently discovered DTLS and QUIC, of which DTLS appear to be most interesting as a "general use" thing. We do of course not actually have a shortage of protocols, but it'd be nice to have something that was a little more suited for the "new" Internet. Full authentication/encryption, something lighter than TCP for multimedia -- and something that actually has a hope of crossing cheap DSL routers.

QUIC looks interesting, but a) is very concentrated on being a transport for http/2, b) is in flux, and c) Google apparently haven't donated any code that makes sense for production (excepting client-side in Chromium).

DTLS is closely connected with WebRTC, has several implementations (though, apparently no support in Erlang/Elixir yet?) and given the current push behind WebRTC appears to be here to stay.

Honestly, I don't know how interesting "hardware phones" are anymore. As long as one can get something to work over wlan with android, things "should" be real-world deployable. And I don't have 100s of SIP devices, so I don't need to care ;-).

It may turn out that wlan+android+WebRTC+DTLS never achieves the kind of latency needed for good live video/audio... I haven't experimented that much, but hopefully it's "good enough".

At any rate, appears that SIP really is the only interop that currently works. At first glance it does seem a little complex for the use-case of text messages, inconvenient in terms of addressing -- but maybe a core focused on WebRTC/IPv6 with gateways for SIP the way to go.

Either way, I guess as long as no big players want interop, it doesn't really matter. One'll just have to make a solution that works for inter-org/inter-friends communication, with the option to federate with those interesting in joining/building a new network. Doesn't look like it'll be easy to take a bite out of FB Messenger/Google Hangouts/WhatsApp etc.

https://webrtcglossary.com/dtls-srtp/