SIP Core Service - Service Specifications
Download OpenAPI specification:Download
Below is a list of key RFCs that define important standards for SIP functionality. Additionally, supported codecs and transport protocols are listed to guide your integration with iotcomms.io SIP Core Services.
Supported SIP RFCs
RFC 3261 - SIP: Session Initiation Protocol
\listPseudoHeader-0
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
Defines how media negotiation is handled using SDP during SIP call setup.RFC 3311 - SIP UPDATE Method
Describes the SIP UPDATE method, which allows modification of session parameters during an active session.RFC 3323 - SIP Extensions for Privacy
Provides mechanisms for ensuring privacy in SIP communications by hiding headers and message content.RFC 3325 - SIP Asserted Identity
Defines how identity can be asserted in SIP signaling for trusted networks, often used in enterprise and carrier networks.RFC 3326 - SIP Reason Header Field
Specifies the Reason header, used to convey why a SIP request was issued or terminated, enhancing call management.RFC 3428 - SIP Extension for Instant Messaging
Defines how SIP can be used for sending instant messages, expanding its use beyond voice and video.RFC 3581 - Symmetric Response Routing
Describes the use of the \"rport\" parameter to facilitate NAT traversal for SIP responses, ensuring correct routing through NATs.RFC 3711 - Secure Real-time Transport Protocol (SRTP)
Specifies the encryption of media streams, securing voice and video communication.RFC 4028 - SIP Session Timer
Introduces session timers to manage the duration of SIP sessions and ensure active connections remain valid.RFC 4566 - SDP: Session Description Protocol
Specifies the format for describing multimedia communication sessions used in SIP to negotiate voice and video calls.RFC 4961 - Symmetric RTP / RTCP
Defines symmetric RTP and RTCP, crucial for media traversal in NAT environments.RFC 5626 - Managing Client-Initiated Connections in SIP
Describes managing long-lived client-initiated connections, which is useful for devices behind NAT or firewalls.RFC 5764 - DTLS-SRTP
Provides a protocol for secure key negotiation for media streams using Datagram Transport Layer Security (DTLS), commonly used in WebRTC and SIP.RFC 6086 - SIP Response Identity Mechanism
Specifies a mechanism to include authenticated identity in SIP responses to help verify message origin authenticity.RFC 7118 - SIP Over WebSocket
Extends SIP to allow signaling over WebSocket, supporting browser-based SIP communication and WebRTC clients.
Relevant sections of the RFCs are implement to provide the speciefed functions for the services which means that some parts of the RFC may be left out.
Supported Audio and Video Codecs
iotcomms.io SIP Core Services support a variety of codecs for audio and video, ensuring compatibility and high-quality communication across different devices and platforms.Please note that the SIP Protocol itself is media agnostic and for the SIP Proxy and SIP Registrar funcation any codec is supported, and the codec to be used is negotiated end to end per RFC 3264.
The codecs listed below are examples of codecs supported for the transcoding function and use-cases where media is terminated by other iotcomms.io services.
Audio Codecs
PCMA (G.711 A-law)
PCMU (G.711 µ-law)
OPUS
G722
G729
GSM
Video Codecs
VP8
VP9
H.264
Transport Protocols
iotcomms.io supports multiple transport protocols, providing flexibility to adapt to various network setups:SIP over UDP, TCP, TLS
SIP over WebSocket
RTP/UDP
SRTP/UDP
DTLS-SRTP
These RFCs and codec references ensure that your SIP devices and applications are fully compatible with iotcomms.io SIP Core Services, supporting secure, high-quality, and scalable communication.