Feature Roadmap
SIP Signaling
| RFC | Title | Status | Notes |
|---|---|---|---|
| RFC 3261 | SIP: Session Initiation Protocol | Partial | UAC only; REGISTER, INVITE, BYE, and digest authentication over TLS/TCP |
| RFC 5626 | Managing Client-Initiated Connections in SIP | Partial | Double-CRLF keepalive ping/pong (§4.4.1) |
| RFC 8760 | SIP Digest Authentication Using AES-HMAC-SHA2 | Complete | MD5, SHA-256, and SHA-512/256 digest responses |
| RFC 3824 | Using E.164 Numbers with SIP | Planned | Phone number mapping into SIP/ENUM |
| RFC 3966 | The tel URI for Telephone Numbers | Planned | Canonical tel: URI scheme |
| RFC 6116 | The E.164 to URI DDDS Application (ENUM) | Planned | DNS-based E.164 number-to-URI mapping |
Media Transport
| RFC | Title | Status | Notes |
|---|---|---|---|
| RFC 3550 | RTP: A Transport Protocol for Real-Time Applications | Complete | Full RTP packet parsing and per-call multiplexing |
| RFC 3551 | RTP Profile for Audio and Video Conferences | Partial | PCMU (0), PCMA (8), G.722 (9), and Opus (111) payload types |
| RFC 3711 | Secure Real-time Transport Protocol (SRTP) | Complete | AES-CM-128-HMAC-SHA1-80 encryption and authentication |
| RFC 4566 | SDP: Session Description Protocol | Partial | Offer/answer model for audio calls; connection and media lines |
| RFC 4568 | SDP Security Descriptions for Media Streams (SDES) | Complete | Inline SRTP key exchange via a=crypto: |
| RFC 5389 | STUN: Session Traversal Utilities for NAT | Complete | Binding Request/Response with XOR-MAPPED-ADDRESS |
| RFC 7983 | Multiplexing Scheme Updates for SRTP Extension for DTLS | Complete | First-byte demultiplexing of STUN vs. RTP/SRTP |
| RFC 7587 | RTP Payload Format for the Opus Speech and Audio Codec | Complete | Dynamic payload type 111 |
| RFC 3533 | The Ogg Encapsulation Format Version 0 | Partial | Minimal Ogg page writer for Opus audio export |
| RFC 4733 | RTP Payload for DTMF Digits, Telephony Tones, and Telephony Signals | Planned | In-band DTMF over RTP |
IVR and Application Services
| RFC | Title | Status | Notes |
|---|---|---|---|
| RFC 6230 | Media Control Channel Framework | Planned | SIP-based control of external media servers |
| RFC 6231 | IVR Control Package for the Media Control Channel Framework | Planned | Interactive voice response over the media control channel |
| RFC 4458 | SIP URIs for Applications such as Voicemail and IVR | Planned | Standardized SIP URI parameters for voicemail and IVR services |
| RFC 3880 | Call Processing Language (CPL) | Planned | XML language for describing call-handling logic |
Voicemail
| RFC | Title | Status | Notes |
|---|---|---|---|
| RFC 3801 | Voice Profile for Internet Mail – version 2 (VPIMv2) | Planned | Voice mail exchange between servers over Internet mail |
| RFC 4239 | Internet Voice Messaging (IVM) | Planned | Standardized Internet voice message format |
Telephony Routing
| RFC | Title | Status | Notes |
|---|---|---|---|
| RFC 2871 | A Framework for Telephony Routing over IP | Planned | Architectural framework for TRIP |
| RFC 3219 | Telephony Routing over IP (TRIP) | Planned | Inter-domain routing of telephony destinations |
| RFC 5115 | Telephony Routing over IP (TRIP) Attribute for Resource Priority | Planned | Priority service support over TRIP |