The Future of SIP in WebRTC
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WebRTC allows for real-time communications like voice and video directly in web browsers without plugins. While SIP is a signaling protocol used for controlling multimedia sessions in VoIP, WebRTC focuses on media and does not define its own signaling protocol. WebRTC needs a separate signaling server and protocol like SIP to fully operate call setup. This marginalizes SIP's importance somewhat by making communications more accessible to developers and embedding them directly into applications rather than separate services. However, WebRTC and SIP can also benefit each other when used together.
The Future of SIP in WebRTC
- 1. The Future of SIP in WebRTC Real Time Communica.ons on the Web BDNOG5 SHAILA SHARMIN LINK3 TECHNOLOGIES LTD DHAKA, 11 APRIL 2016
- 2. WebRTC : Web Real Time CommunicaLons What WebRTC is : Browsers with interac.ve voice & video communica.ons And data sharing capabili.es No download or plug-in • Easy to write audio/video service • Communica.ons become a feature, not the applica.on • Purpose built for the World Wide Web • Collabora.ve W3C and IETF standardiza.on • Peer 2 Peer
- 4. How WebRTC works? Triangle & Trapezoid Trapezoid Triangle
- 5. SIP and WebRTC are different in their funcLons? SIP stands for Session Ini.a.on Protocol, it is a text-based protocol used in Internet telephony (VoIP) for signaling and controlling mul.media sessions. It’s like the square and rectangle concept; all squares are rectangles, but not all rectangles are squares. SIP can exist without WebRTC, but WebRTC needs a signaling protocol to fully operate. The WebRTC vs. SIP baYle is actually a set of two different baYles going on at once: 1. SIP vs. Signaling Protocol X 2. WebRTC vs. VoIP (Browser vs. PSTN) SIP does signaling. And also defines how media gets handled. WebRTC does media. But WebRTC doesn’t define how signaling is handled. Nor does it care.
- 6. Voice Over Internet Protocol § “VoIP” a Broad term § Grown to encompass mul.media, not just voice § Diverse protocols § some well defined standards, some de-facto, some proprietary § Used in a variety of networks § IPv4, IPv6, Public Internet, Private LANs, etc. UC Business VoIP Fixed Line
- 7. Interworking with tradiLonal VoIP IPv4 / IPv6 Network Media Transport Media Descrip.on Signaling Transport Signaling Protocol DTLS-SRTP STUN ICE TURN RTP-Mux RTP BUNDLE Data Channels MSID UDP SDES-SRTP MSRP Unique Transport per Stream Tradi.onal SDP TCP TLS UDP WebSockets HTTP Undefined SIP XMPP H.323 Codecs Opus VP8 H.264 AMR-WB VoIP and WebRTC Similari.es • Transmission of communica.on data between Users in real-.me • Use RTP, SDP O/A • G.711 • Run over IP networks VoIP and WebRTC differences VoIP uses a mul.tude of variants such as VoIP over DSL/ cable modem, VoWiFi/3G, VoLTE, and Rich Communica.on Suite (RCS), while WebRTC is focused on browser-based communica.ons.
- 8. Let's Talk Signaling- WebRTC does not define a signaling protocol But signaling is required for call setup, WebRTC solu.ons must include a signaling server. WebRTC itself doesn't care how that server implements signaling, but it must exist somewhere in the network -- which brings us to SIP. SIP defines signaling. Session Descrip.on Protocol (SDP), defines media. SIP and SDP work together to create, manage, and tear down media sessions of any type. Signaling was lek out of WebRTC for two good reasons: 1. Different applica.ons may require/prefer different protocols. The WebRTC working group did not want to lock it down to something that may turn out to be inadequate for all its uses. 2. WebRTC runs in a Web browser, and support for signaling would require that Web pages be stateful. This becomes problema.c if signaling is lost each .me a page reloads.
- 9. Three aspects of WebRTC that marginalizing the importance of SIP! 1. WebRTC is all about “dumbing down” communicaKons – making it accessible to a lot more developers than just us VoIP engineers. 2. WebRTC is about embedding communicaKons – changing it from a service into a feature of another service. 3. WebRTC is about killing federaKons – WebRTC is en.cing a silo approach to services. You need comms? Just plug WebRTC in and you’re done. No need to think about interworking with others, connec.ng or federa.ng with more networks .