---
Categories:
- Qt
- KDE
- Network
- Telephony
Description:
C: <p>Soft-phone for making telephone calls using SIP over an IP network.</p><p>Twinkle supports direct
IP phone to IP phone communication or a network using a SIP proxy to route your calls.</p><p>In addition
to making basic voice calls Twinkle provides you the following features regardless of the services
that your VoIP service provider might offer.</p><p>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)</p><p>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</p><p>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)</p><p>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)</p>
de: <p>Software-Telefon, um mittels SIP Telefonate über ein IP-Netzwerk zu führen.</p><p>Twinkle unterstützt
direkte Kommunikation von IP-Telefon zu IP-Telefon oder über ein Netzwerk, das einen SIP-Proxy zur
Weiterleitung der Anrufe nutzt.</p><p>Zusätzlich zum einfachen Telefonieren bietet Twinkle die folgenden
Funktionen, unabhängig davon, ob Ihr VoIP-Dienstleister Ihnen diese Dienstleistungen anbietet.</p><p>2
eingehende Anrufkanäle (Leitungen) mehrere aktive Telefonidentitäten benutzerdefinierte Rufzeichen
Anklopfen Verbindung halten 3-Wege-Konferenzschaltung Stummfunktion Rufumleitung bei Anforderung bedingungslose
Rufumleitung Rufumleitung wenn besetzt Rufumleitung wenn niemand abnimmt Rufumleitungsanfrage ablehnen
blinde Rufweiterleitung Rufweiterleitung nach Gesprächsannahme (betreute Rufweiterleitung) (neu) Rufweiterleitungsanfrage
ablehnen Anruf ablehnen Wahlwiederholung »Nicht stören«-Funktion automatische Rufannahme Benachrichtigung
über Nachrichten in Abwesenheit Zielwahl für Sprachnachrichten benutzerdefinierte Skripte bei Anrufereignissen
z.B. zur zielgerichteten Rufablehnung oder für charakteristische Klingeltöne RFC 2833 DTMF-Ereignisse
In-band-DTMF Out-of-band-DTMF (SIP INFO) STUN-Unterstützung für Netzwerke mit NAT sende NAT-Pakete
zur Verbindungshaltung mit STUN NAT-Unterstützung für statische Netze Meldung verlorener Anrufe Geschichte
der Kommunikationsdatensätze für eingehende, ausgehende, erfolgreiche und fehlgeschlagenen Verbindungen
Unterstützung für DNS SRV automatischer Serverwechsel, wenn ein Server nicht verfügbar ist andere
Anwendungen können einen Anruf mit Twinkle aufbauen, z.B. Anruf aus einem Adressbuch Bild im Systembenachrichtigungsabschnitt
Systembenachrichtigungsmenü zum Aufbau und Annahme eines Anrufs mit ausgeblendetem Twinkle benutzerdefinierte
Anzahl an Konvertierungsvorschriften einfaches Adressbuch Unterstützung für UDP und TCP (neu) zur
Übertragung von SIP Präsenz Sofortnachrichtendienst einfacher Dateiaustausch mittels des Sofortnachrichtendienst
Benachrichtigung über das Verfassen einer Sofortnachricht des Gesprächspartners Befehlszeilenschnittstelle
(CLI)</p><p>VoIP-Sicherheit sichere Sprachübetragung mit ZRTP/SRTP Unterstützung von MD5-Hash-Authentifizierung
für alle SIP-Anfragen Unterstützung von AKAv1-MD5-Hash-Authentifizierung für alle SIP-Anfragen (neu)
Rufnummernunterdrückung</p><p>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)</p><p>Für alle Codecs sind folgende Vorverarbeitungsoptionen verfügbar, um die
Qualität am anderen Ende der Leitung zu verbessern. Lautstärkeanpassung (AGC) (neu) Rauschreduzierung
(neu) Erkennung der Sprachaktivität (VAD) (neu) Echoüberwachung (AEC) [experimentell] (neu)</p>
fr: <p>Soft-phone pour passer des appels téléphoniques en utilisant SIP sur un réseau IP.</p><p>Twinkle
prend en charge la communication directe de téléphone IP à téléphone IP ou un réseau utilisant un
proxy SIP pour acheminer vos appels.</p><p>En plus des appels vocaux de base, Twinkle vous offre les
fonctionnalités suivantes quels que soient les services que votre fournisseur de service VoIP peut
offrir.</p><p>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)</p><p>Sécurité
VoIP Sécurise la communication vocale par ZRTP/SRTP Prise en charge d'authentification par hachage
MD5 pour toutes les requêtes SIP Prise en charge d'authentification par hachage AKAv1-MD5 pour toutes
les requêtes SIP (nouveau) Masquage d'identité</p><p>Codecs audio G.711 A-law (64 kbps de charge utile,
8 kHz de fréquence d’échantillonnage) G.711 u-law (64 kbps de charge utile, 8 kHz de fréquence d’échantillonnage)
GSM (13 kbps de charge utile, 8 kHz de fréquence d’échantillonnage) Speex bande étroite (15.2 kbps
de charge utile, 8 kHz de fréquence d’échantillonnage) Speex bande large (28 kbps de charge utile,
16 kHz de fréquence d’échantillonnage) Speex bande ultra large (36 kbps de charge utile, 32 kHz de
fréquence d’échantillonnage) G.726 (16, 24, 32 ou 40 kbps de charge utile, 8 kHz de fréquence d’échantillonnage)</p><p>Pour
tous les codecs, les options de prétraitement suivantes sont disponibles pour améliorer la qualité
à l'autre bout de la ligne. Contrôle de gain automatique (AGC) (nouveau) Réduction de bruit (nouveau)
Détection d'activité vocale (VAD) (nouveau) Contrôle d'écho acoustique (AEC) [expérimental] (nouveau)</p>
it: <p>Telefono software per fare telefonate usando SIP su IP in rete.</p><p>Twinkle supporta comunicazioni
dirette da telefono IP a telefono IP o ad una rete usando un proxy SIP per instradare le chiamate.</p><p>In
aggiunta alle chiamate voce base, Twinkle fornisce le seguenti funzionalità indipendentemente dagli
eventuali servizi del proprio provider VoIP.</p><p>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)</p><p>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</p><p>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)</p><p>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)</p>
ja: <p>IP ネットワーク上で SIP を用いて電話をかけるためのソフトウェア電話です。</p><p>Twinkle は IP 電話から IP 電話への直接通話や、あなたのコールを送るめに SIP
プロキシを使うネットワークに対応しています。</p><p>基本的な音声コールの作成に加え、Twinkle は VoIP サービスプロバイダが提 供するサービスに関係なく、以下の機能を提供します。</p><p>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)</p><p>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</p><p>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)</p><p>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)</p>
pt: <p>Soft-phone para fazer chamadas telefónicas usando SIP sobre uma rede IP.</p><p>O Twinkle suporta
comunicação direta telefone IP a telefone IP ou numa rede usando um proxy SIP para encaminhar as suas
chamadas.</p><p>Além de fazer chamadas de voz básicas, o Twinkle pode oferecer as seguintes funcionalidades,
independente dos serviços que o seu provedor possa oferecer.</p><p>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)</p><p>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</p><p>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)</p><p>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)</p>
pt_BR: <p>Telefone virtual para fazer ligações telefônicas usando SIP sobre rede IP.</p><p>Twinkle tem
suporte a comunicação direta de telefone IP para telefone IP ou uma rede usando proxy SIP para rotear
chamadas.</p><p>Além de fazer chamadas de voz básicas, Twinkle fornece para você os seguintes recursos
além dos serviços que seu provedor de VoIP pode oferecer.</p><p>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)</p><p>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</p><p>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)</p><p>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)</p>
ru: <p>Программный телефон для осуществления звонков, через IP-сеть.</p><p>Twinkle поддерживает как
прямое подключение с IP-телефона на IP-телефон, так и соединение по сети, используя SIP-прокси для
направления звонка.</p><p>Для осуществления простых голосовых звонков Twinkle предоставляет следующие
возможности, даже если провайдер VoIP этого не поддерживает.</p><p>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)</p><p>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</p><p>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)</p><p>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)</p>
sl: <p>Programski telefon za telefonske klice s SIP preko omrežja IP.</p><p>Twinkle podpira neposredno
sporazumevanje od telefona IP do telefona IP ali omrežja z uporabo posredniškega strežnika SIP za
usmeritev vaših klicev.</p><p>Poleg osnovnih zvokovnih klicev vam Twinkle zagotavlja naslednje zmožnosti
ne glede na storitve, ki jih morda ponuja ponudnik storitve VoIP.</p><p>2 videza klicev (liniji) več
določil dejavnih klicev melodije zvonjenja po meri čakajoči klic zadržani klic 3 smerno usmerjanje
konference nemo preusmeritev klicev na zahtevo brezpogojna preusmeritev klicev preusmeritev klicev
ob zaposlenosti preusmeritev klicev, če ni odgovora zahteva preusmeritve blokiranega klica prenos
slepih klicev ponovi zadnji klic ne moti samodejen odgovor pokazatelj čakajočega sporočila hitra številčnica
glasovne pošte uporabniško določljivi skripti, ki se sprožijo ob klicih na primer podpora selektivnega
zvonenja dogodki RFC 2833 DMTF DMTF v pasu DMTF iz pasu (SIP INFO) podpora STUN za prehod NAT pošiljanje
paketov NAT keep alive pri uporabi STUN prehod NAT skozi statično določne pokazatelj zgrešenih klicev
zgodovina podrobnosti klicev za prihajajoče, odhajajoče, uspešne in zgrešene klice podpora DNS SRV
samodejno preklapljanje za nadomestni strežnik, če strežnik ni dosegljiv drugi programi lahko kličejo
preko Twinkle, na primer kličejo preko imenika ikona sistemske vrstice meni sistemske vrstice za izvir
in sprejem klicev, ko ostane Twinkle skrit uporabniško določljiva pravila pretvorbe enostaven imenik
podpora za UDP in TCP (nov) kot prenos za SIP prisotnost hipno sporočanje enostaven prenos datotek
preko hipnega sporočanja pokazatelj sestavljanja hipnega sporočila vmesnik ukazne vrstice (CLI)</p><p>Varnost
VoIP Varno sporazumevanje z ZRTP/SRTP Podpora izvlečkov overitve MD5 za vse zahteve SIP Podpora izvlečkov
overitce AKAv1-MD5 za vse zahteve SIP (novo) Skrivanje identitete</p><p>Zvočni kodeki G.711 A-law
(64 kbps obremenitev, 8 kHz hitrost vzorčenja) G.711 u-law (64 kbps obremenitev, 8 kHz hitrost vzorčenja)
GSM (13 kbps obremenitev, 8 kHz hitrost vzorčenja) Speex narrow band (15.2 kbps obremenitev, 8 kHz
hitrost vzorčenja) Speex wide band (28 kbps obremenitev, 16 kHz hitrost vzorčenja) Speex ultra wide
band (36 kbps obremenitev, 32 kHz hitrost vzorčenja) G.726 (16, 24, 32 or 40 kbps obremenitev, 8 kHz
hitrost vzorčenja)</p><p>Za vse kodeke so na voljo naslednje možnosti predobdelovanja za izboljšanje
kakovosti daljnega klica. samodejen nadzor glasnosti (AGC) (novo) zmanjšanje šuma (novo) zaznavanje
dejavnosti glasu (VAD) (novo) akustični nadzor odmevov (AEC) [preizkusno] (novo)</p>
uk: <p>Програмний телефон для здійснення дзвінків через IP-мережу.</p><p>Twinkle підтримує як пряме
підключення з IP-телефону на IP-телефон, так і з’єднання через мережу, використовуючи SIP-проксі для
спрямування дзвінка.</p><p>Для здійснення простих голосових дзвінків Twinkle надає наступні можливості,
навіть якщо провайдер VoIP цього не підтримує.</p><p>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)</p><p>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</p><p>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)</p><p>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)</p>
ID: twinkle.desktop
Icon:
cached: twinkle_twinkle48.png
Name:
C: Twinkle
Package: twinkle
Summary:
C: A SIP softphone
Type: desktop-app
