本文内容由阿里云实名注册用户自发贡献，版权归原作者所有，阿里云开发者社区不拥有其著作权，亦不承担相应法律责任。具体规则请查看《
2.3 开源SIP服务器比较
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
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
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
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
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
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”).
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
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
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
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
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
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
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)
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
1) OpenSIPS源代码下载:
用svn down下代码 svn cohttps://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.9 opensips_1_9
2) 安装MySQL
3) OpenSIPS安装
root@ubuntu:cd /home/amaryllis/work/project/opensips/
root@ubuntu:make menuconfig
4) OpenSIPS文件配置
a) 修改配置文件opensipsctlrc:
root@ubuntu:gedit /usr/local/opensips_proxy/etc/opensips/opensipsctlrc
b) 安装数据库:
root@ubuntu:cd /usr/local/opensips_proxy/sbin/
root@ubuntu:./opensipsdbctl create
c) 检查M4是否安装:
apt-get install m4
d) 生成opensips_residential_2013-3-10_22:52:46.cfg文件:
root@ubuntu:cd /usr/local/opensips_proxy/sbin/
root@ubuntu:./osipconfig
5) 设置启动项:
root@ubuntu:cd /home/amaryllis/work/project/opensips/packaging/debian
root@ubuntu:cpopensips.init /etc/init.d/opensips
root@ubuntu:chmod+x /etc/init.d/opensips
root@ubuntu:gedit/etc/init.d/opensips
6) 设置默认项opensips.default:
root@ubuntu:cd /home/amaryllis/work/project/opensips/packaging/debian
root@ubuntu:cp opensips.default /etc/default/
root@ubuntu:cd /etc/default/
root@ubuntu:mv opensips.default opensips
root@ubuntu:gedit opensips
7) 启动OpenSIPS:
root@ubuntu:/etc/init.d/opensips restart（重启）
root@ubuntu:/etc/init.d/opensips start（启动）
