Bug#738469: RFA: twinkle -- Voice over Internet Protocol (VoIP) SIP Phone
Package: wnpp
Severity: normal
I request an adopter for the twinkle package.
The package description is:
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)
Description-md5: e13a5ad5bec99b1338d6a8de9eb67858
Homepage: http://www.twinklephone.com/
Section: comm
Priority: optional
Filename: pool/main/t/twinkle/twinkle_1.4.2-4_amd64.deb
Size: 698384
MD5sum: 745ec08fb8f3215d7d011785516eaf9f
SHA1: 6974cd31af049a7d9ed2b7dbce2c2bc6e696a90e
SHA256: 2f0e684885ea2b80936a768bf16445d25fe21e7aec9b77a568eb8e5152df547d
Package: twinkle
Priority: optional
Section: comm
Installed-Size: 4640
Maintainer: Debian VoIP Team <pkg-voip-maintainers@lists.alioth.debian.org>
Architecture: amd64
Source: twinkle (1:1.4.2-2)
Version: 1:1.4.2-2+b2
Depends: libasound2 (>> 1.0.18), libboost-regex1.42.0 (>= 1.42.0-1), libc6 (>= 2.9), libccgnu2-1.7-0, libccrtp1-1.7-0, libgcc1 (>= 1:4.1.1), libgsm1 (>= 1.0.13), libmagic1, libncurses5 (>= 5.7+20100313), libqt3-mt (>= 3:3.3.8b), libreadline6 (>= 6.0), libsndfile1 (>= 1.0.20), libspeex1 (>= 1.2~beta3-1), libspeexdsp1 (>= 1.2~beta3.2-1), libstdc++6 (>= 4.4.0), libx11-6 (>= 0), libxext6, libxml2 (>= 2.7.4), libzrtpcpp-1.4-0, zlib1g (>= 1:1.1.4)
Filename: pool/main/t/twinkle/twinkle_1.4.2-2+b2_amd64.deb
Size: 1758644
MD5sum: 8537211787283bd9dcac949b90f215a7
SHA1: d1bdb30da29e44a2e57057e8a11b303745d37854
SHA256: d480881ae14fca567dcc8a2df9a34935b46b5fc4a8222ccb73db2d31e0ede7b9
Description-en: 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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