WebRTC to SIP
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VoIP call recording
Presence and IM
SIP client engine
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WebRTC (Web Real-Time Communications) is a new technology implemented in modern
to allow calls from browsers as part of the HTML5 protocol suite. Other use-case includes: video chat, screen sharing and file transfer.
became more and more popular in the last year, replacing other technologies such as NPAPI browsers plugins including web clients implemented in Flash, Java and browser add-ons/plugins.
Mizutech provides cutting edge WebRTC implementation:
-built-in the Mizu
WebRTC to SIP gateway
which can be used with any SIP server or PBX
WebRTC SIP client
library and sip webphone
On the server side we provide a transparent WebRTC stack so there is no any special treatment needed for WebRTC calls. They will show up as regular VoIP calls and conversion to/from SIP is fully transparent including both the signaling and media conversion (WebRTC/DTLS/SRTP, ICE/STUN/TURN).
Unfortunately WebRTC also has a list of disadvantages, some of them are very serious, such as:
-not supported by all browsers (not supported by IE, Edge, Safari)
-doesn’t work in corporate networks (When UDP is usually blocked, because WebRTC media will works only with UDP transport)
-needs protocol conversion on the server (DTLS/SRTP/Websocket/TLS)
-doesn’t support popular VoIP codec’s like G.729
-it ads extra complexity and black-boxed by browser without any chance for developers to change the details
solves these issues by using also other
web VoIP engines
running on client side (with the user’s browser) which can provide native (SIP/RTP) VoIP most of the time for customers seeking for a more trouble-free VoIP experience. WebRTC is also fully supported and used when there is no any better technology to make VoIP calls (This depends on client operating system and browser type/version).
More details about WebRTC can be found
WebRTC on the client side can be implemented using low level
or you can use a higher level implementation such as
Copyright (c) 2016 Mizutech S.R.L.