Is SIP the new WebRTC infrastructure?
SIP celebrates 10 years this summer. When it was created, it was created to become the new infrastructure for Internet-wide realtime communication and collaboration. Based on domains, like e-mail, SIP was built for the Internet. But neither the Internet nor corporate networks was ready for it. Many problems was discovered, like NAT traversal, firewall handling and the quality of the existing networks. They where built for store-and-forward networking, getting a file, sending a file – but not realtime communication.
10 years later we see that most of the SIP installations ended up as PSTN-over-IP solutions. SIP addresses are equal to E.164 phone numbers and domain based federations across the Internet is rarely used. We have ended up in a world with VoIP islands that have no interoperability that call each other over the old switch phone network. That was not the vision, that was not the idea driving the development of SIP.
Webrtc- where all the fun happens today!
WebRTC – joint development by W3C and the IETF
Will WebRTC make SIP a dying system?
There will be millions of WebRTC applications, from games to business communication. As soon as you want to communicate across server borders, there is still the issue of finding the other person and setting up a communication session. You need a protocol to do this. WebRTC only involves setting up the media and describing media capabilities – but there still needs to be a way to exchange these attributes in order to set up a session. If the users are using the same web application, the application will be able to manage the exchange of media attributes. This will work for many applications that wants to enhance the existing service with realtime audio and video. As soon as you want to scale beyond that and communicate with users somewhere else, on a SIP phone or a PSTN device, you need a protocol that opens up for reaching out and setting up a session. SIP is exactly such a protocol, a protocol that operates on a session – finds the other party (or multiple users), sets up the session, manage the session and end the session. SIP and WebRTC are perfect buddies. From a SIP standpoint, there’s not a lot of difference between using the multimedia system in a computer than using it in a browser. From the users standpoint, they’ve already installed the browser. It’s there and simple to use.
SIP with WebRTC fulfills the SIP vision
SIP and WebRTC have the possibility of bringing SIP back to it’s root. We can now build a federated global network, like E-mail. And add security that did not exist in e-mail, so that we can trust the network and use it everywhere we connect our devices. There are already many solutions that will take you beyond telephony-style number-based calls, like the Kamailio SIP proxy and the Zoiper and Blink soft phones that support SIP presence and instant messaging.
Edvina stays on top of these new technologies and are happy to help new and existing customers to navigate in this new communications world!
Tags: ietf, kamailio, sip, w3c, webrtc