名称

编程语言

操作系统

许可协议

功能

Asterisk

C

跨平台

GNU GPL/Proprietary

Voice mail

Conference calling

Interactive voice response

automatic call distribution

Cipango

Java

Apache 2.0

Free Switch

C

跨平台

MPL

会议，使用XML-RPC 控制呼叫，Interactive voice response (IVR), TTS/ASR (语音合成和语音识别), Public switched telephone network (PSTN) 接口，可连接模拟和数字中继，VoIP 协议包括 SIP，SCCP, H.323, XMPP, GoogleTalk, t.38 等等

GNU SIP Witch

C++

跨平台

GPL

Call forwarding, call distribution, call hold, presence information and (text) messages, supports encrypted calls and also enables NAT traversal to establish the peer-to-peer connections.

Mobicents

Java

跨平台

LGPL

Mysipswitch

C#

BSD

SIP account creation

Setting up a customized dial plan

Setting up 3rd party SIP Registrations

SIP traffic forwarding

SIP Accounts activity monitoring via the website

SIP traffic monitoring via telnet

Online switchboard: call hold/resume, call transfer/forward, call hangup

Usual security features

Click to Call (Beta)

Possibility to run it on a local computer

Multiple call forwarding

RUBY Dial plans

ENUM Lookup

Opensip

C

Linux, FreeBSD, Solaris

GNU GPL

SIP registrar server

SIP router / proxy (lcr, dynamic routing, dialplan features)

SIP redirect server

SIP presence agent

SIP back-to-back User Agent

SIP IM server (chat and end-2-end IM)

SIP to SMS gateway (bidirectional)

SIP to XMPP gateway for presence and IM (bidirectional)

SIP load-balancer or dispatcher

SIP front end for gateways/asterisk

SIP NAT traversal unit

SIP application server

SailFin

Java

Cross-platform

SIP Express Router

C

Linux, BSD, Solaris

GPL

RFC 3261 functionality, a variety of database backends (mysql, oracle, postgres, radius, text-db), management features (remote management via XML-RPC, load-balancing), NATi traversal, telephony features (LCR, speeddial), multidomain hosting, ENUM, presence, and even more

sipX

Fedora CentOS RHEL

Affero General Public License

traditional private branch exchange (PBX) like voice mail, interactive voice response systems, auto attendants and the like. Furthermore it integrates with Exchange 2007and Active Directory Environments.

Yate

C++

Cross-platform

GNU General Public License with linking exception

VoIP server

SS7 switch

VoIP client

Jabber server

Jabber client

Conference server - with up to 200 voice channels in a single conference

VoIP to PSTN gateway

PC2Phone and Phone2PC gateway

IP Telephony server and/or client

H.323 gatekeeper

H.323 multiple endpoint server

H.323<->SIP Proxy

SIP session border controller

SIP router

SIP registration server

IAX server and/or client

Jingle client or server

MGCP server (Call Agent)

ISDN passive and active recorder

ISDN, RBS, analog passive recorder

Call center server

IVR engine

Prepaid and/or postpaid cards system

YXA

Erlang

Cross-platform

New BSD license

名称

编程语言

操作系统

许可协议

功能

Blink

Python

Mac OS X, Windows and Linux

GNU GPL

OSX Integration (iCloud, iTunes, Address Book, Keychain, Voice Over)

iCloud synchronization for accounts

History menu for outgoing and incoming calls

History browser

System Address Book external plugin (can dial with Blink from Address Book)

Answering machine

Call transfer

Call recording

LDAP directory

Launch external application on incoming calls

Phone number translations

Ekiga

C C++

Unix-like, Windows

GNU General Public License

Call forwarding on busy, no answer, always (SIP and H.323)

Call transfer (SIP and H.323)

Call hold (SIP and H.323)

DTMF support (SIP and H.323)

Basic instant messaging (SIP)

Text chat (SIP and H.323)

Register with several registrars (SIP) and gatekeepers (H.323) simultaneously

Ability to use an outbound proxy (SIP) or a gateway (H.323)

Message waiting indications (SIP)

Audio and video (SIP and H.323)

STUN support (SIP and H.323)

LDAP support

Audio codec algorithms: iLBC, GSM 06.10, MS-GSM, G.711 A-law, G.711 µ-law, G.726, G.721, Speex, G.722, CELT (also G.723.1, G.728, G.729, GSM 06.10, GSM-AMR, G.722.2 [GSM‑AMR-WB] using Intel IPP)

Video codec algorithms: H.261, H.263+, H.264, Theora, MPEG-4

Empathy

C

BSD, Linux, Other Unix-like

GNU GPL

ulti-protocol: Google Talk (Jabber/XMPP), MSN, IRC, Salut, AIM, Facebook, Yahoo!, Gadu Gadu, Groupwise, ICQ and QQ. (Supported protocols depend on installed Telepathy Connection Manager components.) Supports all protocols supported by Pidgin.

File transfer for XMPP, and local networks.

Voice and video call using SIP, XMPP and Google Talk.

Some IRC support.

For detailed list of supported protocol features see here

Conversation theming (see list of supported Adium themes).

Sharing and viewing location information.

Private and group chat (with smileys and spell checking).

Conversation logging.

Automatic away and extended away presence.

Automatic reconnection using Network Manager.

Python bindings for libempathy and libempathy-gtk

Support for collaborative applications (“tubes”).

Jitsi

Java

Linux, Mac OS X, Windows (all Java supported)

LGPL

Attended and blind call transfer

Auto away

Auto re-connect

Auto answer and Auto Forward

Call recording

Call encryption with SRTP and ZRTP

Conference calls

Direct media connection establishment with the ICE protocol

Desktop Streaming

Encrypted password storage using a master password

File transfer for XMPP, AIM/ICQ, Windows Live Messenger, YIM

Instant messaging encryption with OTR

IPv6 support for SIP and XMPP

Media relaying with the TURN protocol

Message Waiting Indication (RFC 3842)

Voice and video calls for SIP and XMPP using H.264 and H.263 or VP8 for video encoding

Wideband audio with SILK, G.722, Speex and Opus

DTMF support with SIP INFO, RTP (RFC 2833/RFC 4733), In-band

Zeroconf via mDNS/DNS-SD (à la Apple's Bonjour)

DNSSEC

Group video support (Jitsi Videobridge)

Packet loss concealment with the SILK and Opus codecs

KPhone

C++

Linux

GPL

Multiple parallel sessions (in the case of audio, one may be active, the others are held).

Own ring tones or "ring music"

NAT-traversal and STUN support

Supported sound systems: ALSA and OSS

SRTP encryption for voice

Presence information

Call Hold

Call transfer

Call forwarding

Auto Answer

Linphone

C

Cross-platform

GNU GPL version 2

SIP user agent compliant with RFC 3261

SIP/UDP, SIP/TCP, SIP/TLS

Supports IPv6

Digest authentication

Supports multiple calls simultaneously with call management features: hold on with music, resume, transfer...

Multiple SIP proxy support: registrar, proxies, outbound proxies

Text instant messaging with delivery notification

Presence using the SIMPLE standard in peer to peer mode

DTMF (telephone tones) support using SIP INFO or RFC 2833

MicroSIP

C/C++

Windows

GNU General Public License

Profile of a lightweight background application[2]

Small memory footprint (<20 mb RAM usage)

Strong adherence to the SIP standard

Support for a number of codecs: Speex (narrow band and wideband), G.711 (u-law, a-law), GSM, iLBC, SILK (narrow band, wideband and ultra wideband), G.722

No Support for VP8 codec as of now

STUN and ICE NAT traversal

SIP SIMPLE presence and messaging

QuteCom

C/C++

Cross-platform

GNU General Public License

SIP compliance

Provider agnostic

Allows users to send SMS to France

NAT traversal

Cross-platform

Audio smileys

Qt-based GUI

Chatting with MSN, AIM, ICQ, Yahoo and XMPP users

Encryption via SRTP, but key exchange over Everbee key that is not a Standard

Uses standard Session Initiation Protocol

SFLphone

C / C++

Linux

GNU General Public License 3

SIP and IAX compatible

Unlimited number of calls

Call recording

Attended call transfer

Call hold

Multiple audio conferencing (from 0.9.7 version)

TLS and ZRTP support (from 0.9.7 version)

Audio codecs supported: G711u, G711a, GSM, Speex (8, 16, 32 kHz), CELT, G.722

Multiple SIP accounts support

STUN support per account (0.9.7)

DTMF support (SIP INFO)

Instant messaging

Call history + search feature

Silence detection with Speex audio codec

Account assistant wizard

Central server providing free SIP/IAX account

SIP presence subscription

Video multiparty conferencing (EXPERIMENTAL)

Multichannel audio support [EXPERIMENTAL]

Flac and OGG/Vorbis ringtone support

Desktop notification: voicemail number, incoming call, information messages

Minimize on start-up

Minimize to tray

not Direct IP-to-IP SIP call - P2P is not supported by IAx2 according to the documentation

SIP Re-invite

Address book support: Evolution Data Server integration (for the GNOME client), KABC integration for the KDE client

PulseAudio support

Native ALSA interface, DMix support

Locale settings: French, English, Russian, German, Chinese, Spanish, Italian, Vietnamese

Automatic opening of incoming URL

Telephone

Objective-C

Mac OS X

BSD License

Twinkle

C++

GNU/Linux

GNU General Public License

2 call appearances (lines)

Multiple active call identities

Custom ring tones

Call Waiting

Call Hold

3-way conference calling

Mute

Call redirection on demand

Call redirection unconditional

Call redirection when busy

Call redirection no answer

Reject call redirection request

Blind call transfer

Call transfer with consultation (attended call transfer)

Reject call transfer request

Call reject

Repeat last call

Do not disturb

Auto answer

Message Waiting Inidication

Voice mail speed dial

User defineable scripts triggered on call events

E.g. to implement selective call reject or distinctive ringing

RFC 2833 DTMF events

Inband DTMF

Out-of-band DTMF (SIP INFO)

STUN support for NAT traversal

Send NAT keep alive packets when using STUN

NAT traversal through static provisioning

Persistent TCP connections for NAT traversal

Missed call indication

History of call detail records for incoming, outgoing, successful and missed calls

DNS SRV support

Automatic failover to an alternate server if a server is unavailable

Other programs can originate a SIP call via Twinkle, e.g. call from address book

System tray icon

System tray menu to quickly originate and answer calls while Twinkle stays hidden

User defineable number conversion rules

Simple address book

Support for UDP and TCP as transport for SIP

Presence

Instant messaging

Simple file transfer with instant message

Instant message composition indication

Command line interface (CLI)

Yate

C++

Cross-platform

GNU General Public License with linking exception

VoIP server

SS7 switch

VoIP client

Jabber server

Jabber client

Conference server - with up to 200 voice channels in a single conference

VoIP to PSTN gateway

PC2Phone and Phone2PC gateway

IP Telephony server and/or client

H.323 gatekeeper

H.323 multiple endpoint server

H.323<->SIP Proxy

SIP session border controller

SIP router

SIP registration server

IAX server and/or client

Jingle client or server

MGCP server (Call Agent)

ISDN passive and active recorder

ISDN, RBS, analog passive recorder

Call center server

IVR engine

Prepaid and/or postpaid cards system