packages [ lucid ]
[ precise ]
[ 源代码: twinkle ]
软件包: twinkle (1:1.4.2-2build1) [universe]
twinkle 的相关链接
Ubuntu 的资源:
下载源码包 twinkle:
维护者:
Please consider filing a bug or asking a question via Launchpad before contacting the maintainer directly.
Original Maintainers (usually from Debian):
- Debian VoIP Team (Mail Archive)
- Kilian Krause
- Jose Carlos Garcia Sogo
- Mark Purcell
- Santiago Garcia Mantinan
- Mikael Magnusson
It should generally not be necessary for users to contact the original maintainer.
外部的资源:
- 主页 [www.twinklephone.com]
相似软件包:
Voice over Internet Protocol (VoIP) SIP Phone
Soft-phone for making telephone calls using SIP over an IP network.
Twinkle supports direct IP phone to IP phone communication or a network using a SIP proxy to route your calls.
In addition to making basic voice calls Twinkle provides you the following features regardless of the services that your VoIP service provider might offer.
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) (new) Reject call transfer request Call reject Repeat last call Do not disturb Auto answer Message Waiting Indication Voice mail speed dial User definable scripts triggered on call events E.g. to implement selective call reject or distinctive ringing RFC 2833 DTMF events In-band 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 Missed call indication History of call detail records for incoming, outgoing, successful and missed DNS SRV support Automatic fail-over to an alternate server if a server is unavailable Other programs can originate a call via Twinkle, e.g. call from address book System tray icon System tray menu to originate and answer calls while Twinkle stays hidden User definable number conversion rules Simple address book Support for UDP and TCP (new) as transport for SIP Presence Instant messaging Simple file transfer with instant message Instant message composition indication Command line interface (CLI)
VoIP security
Secure voice communication by ZRTP/SRTP MD5 digest authentication support for all SIP requests AKAv1-MD5 digest authentication support for all SIP requests (new) Identity hiding
Audio codecs
G.711 A-law (64 kbps payload, 8 kHz sampling rate) G.711 u-law (64 kbps payload, 8 kHz sampling rate) GSM (13 kbps payload, 8 kHz sampling rate) Speex narrow band (15.2 kbps payload, 8 kHz sampling rate) Speex wide band (28 kbps payload, 16 kHz sampling rate) Speex ultra wide band (36 kbps payload, 32 kHz sampling rate) G.726 (16, 24, 32 or 40 kbps payload, 8 kHz sampling rate)
For all codecs the following preprocessing options are available to improve quality at the far end of a call.
Automatic gain control (AGC) (new) Noise reduction (new) Voice activity detection (VAD) (new) Acoustic echo control (AEC) [experimental] (new)
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