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Cloud SIP UA Service - Service Specifications

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Service Specifications

The Cloud SIP UA Service provides SIP communication for lightweight devices that require efficient, scalable, and secure integration with SIP systems. The service supports key industry standards and provides seamless compatibility with diverse SIP ecosystems.

Supported SIP RFCs

The Cloud SIP UA Service complies with the following SIP standards to ensure 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 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.

Supported Codecs

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

  • Audio Codecs

    • PCMA (G.711 A-law)

      Standard narrowband codec used in telephony.

    • PCMU (G.711 µ-law)

      A widely adopted codec in North America for VoIP.

    • OPUS

      A versatile codec supporting narrowband to fullband audio, ideal for high-quality applications.

    • G.722

      A wideband codec offering better audio quality than traditional narrowband codecs.

    • G.729

      A low-bandwidth codec often used in constrained network environments.

    • GSM

      A codec for mobile systems offering efficient compression.

Transport Protocols

To ensure secure and reliable communication, the Cloud SIP UA Service supports the following transport protocols:

  • SIP over UDP, TCP, and TLS

    Provides flexible connection options to meet different network requirements.

  • RTP/UDP and SRTP/UDP

    Handles media transport with or without encryption for compatibility with legacy SIP systems.

Security Features

The Cloud SIP UA Service incorporates robust security measures to safeguard communication:

  • TLS Encryption for SIP Signaling

    Ensures secure SIP communication.

  • SRTP for Media Streams

    Encrypts RTP media streams to protect audio data.

  • Secure Authentication

    Uses API keys and secure methods for device authentication.

Integration Capabilities

  • SIP Core Integration

    Enables trunk connectivity to external systems, including PSTN, and supports codec transcoding for compatibility.

  • SIP Mediaserver Integration

    Provides advanced call functionalities like IVR, media playback, and real-time media processing.

These specifications ensure that the Cloud SIP UA Service delivers a secure, compatible, and scalable communication solution, tailored for real-time applications and lightweight devices.