.gif)
Twinkle (software)
Encyclopedia
Twinkle is a free
and open source software application for Voice over Internet Protocol (VoIP) voice communications in IP networks, such as the Internet
. It is designed for GNU/Linux operating systems and uses the Qt toolkit for its graphical user interface. For call signaling it employs the Session Initiation Protocol
(SIP). It also features direct IP-to-IP calls. Media streams are transmitted via the Real-time Transport Protocol
(RTP) which may be encrypted with the Secure Real-time Transport Protocol
(SRTP) and the ZRTP
security protocols.
Since version 1.3.2 (September 2008), Twinkle supports message exchange and a buddy-list feature for presence notification, showing the online-status of predefined communications partners (provider-support needed).
Free software
Free software, software libre or libre software is software that can be used, studied, and modified without restriction, and which can be copied and redistributed in modified or unmodified form either without restriction, or with restrictions that only ensure that further recipients can also do...
and open source software application for Voice over Internet Protocol (VoIP) voice communications in IP networks, such as the Internet
Internet
The Internet is a global system of interconnected computer networks that use the standard Internet protocol suite to serve billions of users worldwide...
. It is designed for GNU/Linux operating systems and uses the Qt toolkit for its graphical user interface. For call signaling it employs the Session Initiation Protocol
Session Initiation Protocol
The Session Initiation Protocol is an IETF-defined signaling protocol widely used for controlling communication sessions such as voice and video calls over Internet Protocol . The protocol can be used for creating, modifying and terminating two-party or multiparty sessions...
(SIP). It also features direct IP-to-IP calls. Media streams are transmitted via the Real-time Transport Protocol
Real-time Transport Protocol
The Real-time Transport Protocol defines a standardized packet format for delivering audio and video over IP networks. RTP is used extensively in communication and entertainment systems that involve streaming media, such as telephony, video teleconference applications, television services and...
(RTP) which may be encrypted with the Secure Real-time Transport Protocol
Secure Real-time Transport Protocol
The Secure Real-time Transport Protocol defines a profile of RTP , intended to provide encryption, message authentication and integrity, and replay protection to the RTP data in both unicast and multicast applications...
(SRTP) and the ZRTP
ZRTP
ZRTP is a cryptographic key-agreement protocol to negotiate the keys for encryption between two end points in a Voice over Internet Protocol phone telephony call based on the Real-time Transport Protocol. It uses Diffie-Hellman key exchange and the Secure Real-time Transport Protocol for...
security protocols.
Since version 1.3.2 (September 2008), Twinkle supports message exchange and a buddy-list feature for presence notification, showing the online-status of predefined communications partners (provider-support needed).
Supported audio formats
- G.711G.711G.711 is an ITU-T standard for audio companding. It is primarily used in telephony. The standard was released for usage in 1972. Its formal name is Pulse code modulation of voice frequencies. It is required standard in many technologies, for example in H.320 and H.323 specifications. It can also...
A-law: 64 kbit/s payload, 8 kHz sampling rate - G.711 μ-law: 64 kbit/s payload, 8 kHz sampling rate
- G.726G.726G.726 is an ITU-T ADPCM speech codec standard covering the transmission of voice at rates of 16, 24, 32, and 40 kbit/s. It was introduced to supersede both G.721, which covered ADPCM at 32 kbit/s, and G.723, which described ADPCM for 24 and 40 kbit/s. G.726 also introduced a new...
: 16, 24, 32 or 40 kbit/s payload, 8 kHz sampling rate - GSM: 13 kbit/s payload, 8 kHz sampling rate
- iLBCILBCInternet Low Bitrate Codec is an open source royalty-free narrowband speech codec, developed by Global IP Solutions formerly Global IP Sound . It was formerly licensed as a freeware with limited commercial use, but since 2011 it is available under an open source license as a part of the open...
: 13.3 or 15.2 kbit/s payload, 8 kHz sampling rate - SpeexSpeexSpeex is a patent-free audio compression format designed for speech and also a free software speech codec that may be used on VoIP applications and podcasts. It is based on the CELP speech coding algorithm. Speex claims to be free of any patent restrictions and is licensed under the revised BSD...
narrow band: 15.2 kbit/s payload, 8 kHz sampling rate - Speex wide band: 28 kbit/s payload, 16 kHz sampling rate
- Speex ultra wide band: 36 kbit/s payload, 32 kHz sampling rate
See also
- Comparison of VoIP softwareComparison of VoIP softwareVoIP software is used to conduct telephone-like voice conversations across Internet Protocol based networks. VoIP stands for "Voice over IP". For residential markets, VoIP phone service is often cheaper than traditional public switched telephone network service and can remove geographic...
- List of SIP software
- Opportunistic encryptionOpportunistic encryptionOpportunistic Encryption refers to any system that, when connecting to another system, attempts to encrypt the communications channel otherwise falling back to unencrypted communications. This method requires no pre-arrangement between the two systems.Opportunistic encryption can be used to...