Wiki -VoIP Topics

WebRTC-SIP gateway definition

WebRTC-SIP gateways translate the signaling and media between WebRTC and SIP endpoints to allow interconnections between these protocols.

  • WebRTC is the abbreviation for Web Real-Time Communication and means a collection of API and protocols allowing real-time collaboration for web browsers and native WebRTC applications such as voice calls, video calls, file transfer, instant messaging and screen sharing.
  • SIP is the abbreviation for Session Initiation Protocol and means a communications protocol for signaling to control media communication sessions, often used together with RTP to carry the media streams.
A WebRTC gateway core functionality is to provide conversions between these protocols which is required if a party understand only one of these and the other party only the another one. For example:
  • Common browsers has full plugin-free support for WebRTC but no any support for native SIP/RTP (because of the browser sandbox restrictions it is impossible to connect and send packets from a browsers via plan TCP or UDP which is required for SIP/RTP).
  • On the other side, most common VoIP servers (softswitch, IP-PBX) nowadays comes with a robust SIP core but no WebRTC capabilities or their WebRTC support is very poor with various bugs, incompatibilities and sub-optimal NAT/firewall handling which is not suitable for corporate customers.
Thus the need for WebRTC-SIP gateways to convert between these worlds.

webrtc to sip gateway convert

The main tasks performed by a WebRTC-SIP gateway are the followings:

  1. handle SIP signaling from HTTP/TCP based websocket and converts it to plain UDP SIP signaling
  2. handle media from WebRTC in DTLS/SRTP and convert it to plain RTP over UDP
  3. handle ICE negotiations
Optional features of a WebRTC-SIP gateway include the followings:

  1. built-in STUN and TURN servers (otherwise these needs to be installed separately and fine-tuned to match your network needs)
  2. codec transcoding capabilities
  3. intelligent media relay via local UDP and TCP candidates to ensure proper functionality for NAT restricted networks and clients behind firewalls
  4. SBC and security features to protect your network
  5. extra services for features which can’t be directly translated between WebRTC and SIP such as file transfer or conference
  6. monitoring, routing, alerting and other management services

WebRTC-SIP gateway solutions:

There are a few commercial and open-source solutions available with varying quality:

  • MRTC an all-in-one WebRTC to SIP gateway software (with built-in TLS, TURN and media relays)
  • webrtc2sip a modularized solution from Doubango (requires separate TURN server and manual text config)
  • Reve WebRTC from Reve systems (requires sign-up, not tested)
  • squire SBC is an SBC named ABC gateway (but has also WebRTC capabilities)
  • PortSIP webrtc gateway with SIP conversion capabilities (from the company with the same name)
  • Janus is a general purpose SIP gateway (with support for protocols like SCTP and message queues)
External links and resources:

You can find more details about WebRTC-SIP gateways on the below lins:

  • A good description about how a WebRTC-SIP gateway works can be found on the MRTC homepage here
  • A general description about WebRTC gateways (not SIP specific) can be found here.
  • You can find a more technical details about WebRTC-SIP gateways here
Mizutech WebRTC-SIP solutions:

Mizutech offers WebRTC-SIP gateway solutions for both small (small-business) or large (enterprise) networks and also a broad range of other professional WebRTC-SIP software solutions.