193 lines
6.8 KiB
Text
193 lines
6.8 KiB
Text
README
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------
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libre - "Generic library for real-time communications with async IO support"
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Copyright (C) 2010 - 2016 Creytiv.com
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Distributed under BSD license
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Design goals:
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* Portable POSIX source code (ANSI C89 and ISO C99 standard)
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* Robust, fast, low memory footprint
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* RFC compliance
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* IPv4 and IPv6 support
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Modules:
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name: status: description:
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* aes unstable AES (Advanced Encryption Standard)
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* base64 testing Base-64 encoding/decoding functions
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* bfcp unstable The Binary Floor Control Protocol (BFCP)
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* conf testing Configuration file parser
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* crc32 testing 32-bit CRC defined in ITU V.42
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* dbg testing Debug printing
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* dns stable DNS resolving (NAPTR, SRV, A)
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* fmt testing Formatted printing and regular expression
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* hash testing Hashmap table
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* hmac testing HMAC: Keyed-Hashing for Message Authentication
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* http unstable HTTP parser (RFC 2616)
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* httpauth testing HTTP-based Authentication (RFC 2617)
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* ice unstable Interactive Connectivity Establishment (ICE)
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* jbuf testing Jitter buffer
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* json unstable JavaScript Object Notation (JSON)
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* list stable Sortable doubly-linked list handling
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* lock testing Resource locking functions
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* main testing Main poll loop
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* mbuf stable Linear memory buffers
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* md5 stable The MD5 Message-Digest Algorithm (RFC 1321)
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* mem stable Memory referencing
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* mod testing Run-time module loading
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* mqueue testing Thread-safe message queue
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* msg unstable Generic message component library
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* natbd unstable NAT Behavior Discovery using STUN
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* net testing Networking routines
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* odict unstable Ordered Dictionary
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* rtp testing Real-time Transport Protocol
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* sa stable Socket Address functions
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* sdp testing Session Description Protocol
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* sha testing Secure Hash Standard, NIST, FIPS PUB 180-1
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* sip stable Core SIP library
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* sipevent testing SIP Event framework
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* sipreg stable SIP register client
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* sipsess stable SIP Sessions
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* srtp unstable Secure Real-time Transport Protocol (SRTP)
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* stun stable Session Traversal Utilities for NAT (STUN)
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* sys testing System information
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* tcp testing TCP transport
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* telev testing Telephony Events (RFC 4733)
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* tls unstable Transport Layer Security
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* tmr stable Timer handling
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* turn stable Obtaining Relay Addresses from STUN (TURN)
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* udp testing UDP transport
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* uri testing Generic URI library
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* websock unstable WebSocket Client and Server
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legend:
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"stable" - Code complete; Stable code and stable API
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"testing" - Code complete, but API might change
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"unstable" - Code complete but not completely tested
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"development" - Code is under development
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Features:
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* RFC 1321 - The MD5 Message-Digest Algorithm
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* RFC 1886 - DNS Extensions to support IP version 6
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* RFC 2032 - RTP Payload Format for H.261 Video Streams
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* RFC 2616 - Hypertext Transfer Protocol -- HTTP/1.1
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* RFC 2617 - HTTP Authentication: Basic and Digest Access Authentication
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* RFC 2782 - A DNS RR for Specifying the Location of Services (DNS SRV)
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* RFC 2915 - The Naming Authority Pointer (NAPTR) DNS Resource Record
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* RFC 3261 - SIP: Session Initiation Protocol
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* RFC 3263 - Locating SIP Servers
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* RFC 3264 - An Offer/Answer Model with SDP
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* RFC 3265 - SIP-Specific Event Notification
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* RFC 3327 - SIP Extension Header Field for Registering Non-Adjacent Contacts
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* RFC 3428 - SIP Extension for Instant Messaging
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* RFC 3489 - STUN - Simple Traversal of UDP Through NATs
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* RFC 3515 - The SIP Refer Method
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* RFC 3550 - RTP: A Transport Protocol for Real-Time Applications
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* RFC 3551 - RTP Profile for Audio and Video Conferences with Minimal Control
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* RFC 3555 - MIME Type Registration of RTP Payload Formats
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* RFC 3556 - SDP Bandwidth Modifiers for RTCP Bandwidth
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* RFC 3581 - An Extension to SIP for Symmetric Response Routing
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* RFC 3605 - RTCP attribute in SDP
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* RFC 3711 - The Secure Real-time Transport Protocol (SRTP)
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* RFC 3969 - The IANA URI Parameter Registry for SIP
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* RFC 3994 - Indication of Message Composition for Instant Messaging
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* RFC 4346 - The TLS Protocol Version 1.1
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* RFC 4566 - SDP: Session Description Protocol
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* RFC 4582 - The Binary Floor Control Protocol (BFCP)
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* RFC 4585 - Extended RTP Profile for RTCP-Based Feedback
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* RFC 4733 - RTP Payload for DTMF Digits, Telephony Tones, and Teleph. Signals
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* RFC 4961 - Symmetric RTP / RTP Control Protocol (RTCP)
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* RFC 5118 - SIP Torture Test Messages for IPv6
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* RFC 5245 - Interactive Connectivity Establishment (ICE)
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* RFC 5389 - Session Traversal Utilities for NAT (STUN)
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* RFC 5626 - Managing Client-Initiated Connections in SIP
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* RFC 5761 - Multiplexing RTP Data and Control Packets on a Single Port
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* RFC 5766 - Traversal Using Relays around NAT (TURN)
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* RFC 5768 - Indicating Support for ICE in SIP
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* RFC 5769 - Test vectors for STUN
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* RFC 5780 - NAT Behaviour Discovery Using STUN
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* RFC 6026 - Correct Transaction Handling for 2xx Resp. to SIP INVITE Requests
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* RFC 6156 - TURN Extension for IPv6
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* RFC 6188 - The Use of AES-192 and AES-256 in Secure RTP
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* RFC 6455 - The WebSocket Protocol
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* RFC 7159 - JavaScript Object Notation (JSON)
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* RFC 7350 - DTLS as Transport for STUN
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* Symmetric RTP
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* draft-ietf-bfcpbis-rfc4582bis-08
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Supported platforms:
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* Linux
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* FreeBSD
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* OpenBSD
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* NetBSD
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* Symbian OS (not supported)
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* Solaris
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* Windows
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* Apple Mac OS X and iOS
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* Android
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Supported compilers:
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* gcc (v2.9x to v4.x)
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* gcce
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* ms vc2003 compiler
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* clang
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Feedback:
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- Please send feedback to <libre [at] creytiv.com>
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Coding guidelines:
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* Use enum for constants where appropriate
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* Use const as much as possible (where appropriate)
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* Use C99 data types (intN_t, uintN_t, bool)
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* Hide data-types in .c files where possible (use struct foo)
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* Avoid malloc/free, use mem_alloc/mem_deref instead
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* CVS/svn/git tags are NOT allowed in the code!
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* Avoid bit-fields in structs which are not portable
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* Use dummy handlers for timing-critical callbacks
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* return err, return alloced objects as pointer-pointers
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* in allocating functions, first arg is always double pointer
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* Use POSIX error-codes; EINVAL for invalid args, EBADMSG for
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parse errors and EPROTO for protocol errors
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Transport protocols:
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TCP UDP TLS DTLS
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~~~ ~~~ ~~~ ~~~~
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BFCP - yes - -
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DNS yes yes - -
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HTTP yes n/a yes n/a
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ICE - yes - -
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RTP - yes - -
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RTCP - yes - -
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SIP yes yes yes -
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STUN yes yes yes yes
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TURN yes yes yes yes
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WEBSOCK yes n/a yes n/a
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