1 Images Source: Google Images WebRTC and VoIP: bridging the h>p://es.linkedin.com/in/victorpascualavila
3 Use Cases WebRTC enables innovadve use cases on the Web WebRTC It s not meant to be the new Web Telephony But interworking towards exisdng legacy is required Extend exisdng SIP environments into the excidng new web domain Service Provider subscriber access via WebRTC methods Browser- based RTC to complement SIP offerings or simply extend exisdng SIP- services over web Voice service extension with web- phone New Telco- OTT services (over- the- top) RCS- e (rich communicadons services enhanced) Conferencing Web- based comms provider PSTN break- out Enterprise UC without thick or thin client sow phones Easier to maintain & break single UC vendor lock Contact centers embedding RTC into customer service web pages Customer sadsfacdon & lower costs Not only Web browsers (e.g. Chrome, Firefox) but also nadve support via apps or OS (e.g. set- top boxes, FirefoxOS)
5 Media Plane A browser- embedded media engine Audio codecs G.711, Opus are MTI Video codecs H.264 vs. VP8 (MTI TBD - IPR discussion) Media codecs are negodated with SDP (for now at least) Best- of- breed echo canceler Video ji>er buffer, image enhancer Requires Secure RTP (SRTP) DTLS Requires Peer- 2- peer NAT traversal tools (STUN, TURN, ICE) trickle ICE MulDplexing: RTPs & RTP+RTCP Yes, your favorite SIP client implementadon is compadble with most of this. But, the vast majority of deployments Use plain RTP Do not support STUN/TURN/ICE Do not support muldplexing (ok, not really an issue) Use different codecs that might not be supported on the WebRTC side
6 OpDon 1: Just ignore the gap And expect all deployed equipment to be upgraded soon
7 OpDon 2: Try to fix it When/While necessary (hopefully with the right soludon!)
8 OK, might need to do a couple of fixes to make things work, but HOW?
9 THE UNMENTIONABLE THE GATEWAY
10 WebRTC Gateway Taxonomy (great blog post by Chad Hart) SIP- over- WebSockets gateway act as a SIP- server and terminate the WebSocket from the browser and convert that to UDP/TCP transport into the SIP network Web API to SIP gateway act as a web- server and convert the API calls to SIP Media controller ICE/STUN/TURN SRTP/RTP interworking Mux/demux Transcoding gateway convert from one codec to another (incl. video!) Flash RTMP gateway majority of browsers on the web today support flash and not WebRTC Several implementadon opdons: one or more of the opdons above WebRTC as a feature/product/service Other features (?): idendty, charging, session recording, LI, middlebox- traversal, etc.
! This is a sample chapter of WebRTC: APIs and RTCWEB Protocols of the HTML5 Real-Time Web by Alan B. Johnston and Daniel C. Burnett, Second Edition. For more information or to buy the paperback or ebook
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