WebRTC Service - Service Specifications

The WebRTC Service is designed to support WebRTC-based communication, integrating seamlessly with SIP systems and ensuring secure, high-quality media transmission. It complies with key industry standards and supports a wide range of codecs and protocols to maintain compatibility with diverse WebRTC and SIP ecosystems.

The WebRTC Service complies with the following SIP standards, ensuring interoperability and reliability:

• RFC 3261 - SIP: Session Initiation Protocol

  Defines the core protocol for initiating, maintaining, and terminating SIP communication sessions.

• RFC 3263 - SIP: Locating SIP Servers

  Describes how SIP clients use DNS to locate SIP servers.

• RFC 3264 - An Offer/Answer Model with SDP

  Provides guidelines for media negotiation using Session Description Protocol (SDP).

• RFC 3311 - SIP UPDATE Method

  Allows modification of session parameters during an active session.

• RFC 3323 - SIP Extensions for Privacy

  Ensures privacy in SIP communications by hiding headers and message content.

• RFC 3325 - SIP Asserted Identity

  Defines identity assertion in SIP signaling for trusted networks.

• RFC 3326 - SIP Reason Header Field

  Adds a header for conveying the reason for SIP request or termination.

• RFC 3711 - Secure Real-Time Transport Protocol (SRTP)

  Secures RTP media streams, ensuring encrypted voice and video transmission.

• RFC 4028 - SIP Session Timer

  Introduces session timers to manage the duration of SIP sessions.

• RFC 4566 - SDP: Session Description Protocol

  Specifies the format for multimedia communication session descriptions used in SIP.

• RFC 5764 - DTLS-SRTP

  Defines secure key negotiation for media streams, widely used in WebRTC.

• RFC 7118 - SIP Over WebSocket

  Extends SIP signaling to WebSocket connections, enabling browser-based SIP communication for WebRTC clients.

The WebRTC Service supports a range of audio and video codecs to ensure compatibility and high-quality communication across various devices:

• Audio Codecs

  • PCMA (G.711 A-law)

    Widely used in telephony, provides standard narrowband audio quality.

  • PCMU (G.711 µ-law)

    Another commonly used codec in VoIP, especially in North America.

  • OPUS

    A versatile codec supporting narrowband to fullband audio, used in WebRTC for high-quality audio.

  • G.722

    Wideband codec offering higher audio quality than standard narrowband codecs.

• Video Codecs

  • VP8

    A WebRTC-supported codec providing efficient, high-quality video compression.

  • H.264

    A widely adopted video codec, compatible with various devices and networks.

To facilitate secure and reliable communication, the WebRTC Service supports the following transport protocols:

• SIP over WebSocket (WSS)

  Listens on wss:// at port 7443, enabling secure browser-based SIP signaling from WebRTC application

• SIP over UDP, TCP, and TLS

  Ensures flexible connection options for various remote system and trunk connectivity

• RTP/UDP and SRTP/UDP

  Supports media transport with or without encryption towards legacy SIP systems

• DTLS-SRTP

  Used for secure key exchange in WebRTC applications, ensuring encrypted media streams.

These Service Specifications ensure that the WebRTC Service provides a secure, compatible, and high-quality communication solution across diverse WebRTC and SIP environments, meeting the demands of real-time applications.