PhonerLite
Encyclopedia
Description
Phoner and PhonerLite are softphoneSoftphone
A softphone is a software program for making telephone calls over the Internet using a general purpose computer, rather than using dedicated hardware. Often a softphone is designed to behave like a traditional telephone, sometimes appearing as an image of a phone, with a display panel and buttons...
applications for Windows operating systems available as freeware
Freeware
Freeware is computer software that is available for use at no cost or for an optional fee, but usually with one or more restricted usage rights. Freeware is in contrast to commercial software, which is typically sold for profit, but might be distributed for a business or commercial purpose in the...
. Phoner is a multiprotocol telephony application supporting telephony via CAPI
Common ISDN Application Programming Interface
The Common ISDN Application Programming Interface is an ISDN-conform standardized software interface. With the help of the CAPI-Interface computer software intended for the use with ISDN can be provided, without knowledge of the deployed, proprietary ISDN card.CAPI was designed from 1989 by German...
, TAPI
Telephony Application Programming Interface
The Telephony Application Programming Interface is a Microsoft Windows API, which provides computer telephony integration and enables PCs running Microsoft Windows to use telephone services. Different versions of TAPI are available on different versions of Windows...
and VoIP, while PhonerLite provides a specialized and optimized user interface
User interface
The user interface, in the industrial design field of human–machine interaction, is the space where interaction between humans and machines occurs. The goal of interaction between a human and a machine at the user interface is effective operation and control of the machine, and feedback from the...
for VoIP only. Beside the different user interface focus both programms share the same code base.
Both programs use 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...
for VoIP call signalisation. Beside of server based infrastructure, direct IP to IP calls are supported too. 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...
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. Phoner provides as well an interface for configuring and using all supplementary ISDN
Integrated Services Digital Network
Integrated Services Digital Network is a set of communications standards for simultaneous digital transmission of voice, video, data, and other network services over the traditional circuits of the public switched telephone network...
services provided via CAPI and thus needs an ISDN terminal adapter hardware installed in the computer.
Phoner and PhonerLite support IPv4
IPv4
Internet Protocol version 4 is the fourth revision in the development of the Internet Protocol and the first version of the protocol to be widely deployed. Together with IPv6, it is at the core of standards-based internetworking methods of the Internet...
and IPv6
IPv6
Internet Protocol version 6 is a version of the Internet Protocol . It is designed to succeed the Internet Protocol version 4...
connections by using UDP
User Datagram Protocol
The User Datagram Protocol is one of the core members of the Internet Protocol Suite, the set of network protocols used for the Internet. With UDP, computer applications can send messages, in this case referred to as datagrams, to other hosts on an Internet Protocol network without requiring...
, TCP
Transmission Control Protocol
The Transmission Control Protocol is one of the core protocols of the Internet Protocol Suite. TCP is one of the two original components of the suite, complementing the Internet Protocol , and therefore the entire suite is commonly referred to as TCP/IP...
and TLS
Transport Layer Security
Transport Layer Security and its predecessor, Secure Sockets Layer , are cryptographic protocols that provide communication security over the Internet...
.
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.722G.722G.722 is a ITU-T standard 7 kHz wideband speech codec operating at 48, 56 and 64 kbit/s. It was approved by ITU-T in November 1988. Technology of the codec is based on sub-band ADPCM ....
: 64 kbit/s payload, 16 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 kbit/s payload, 8 kHz sampling rate - Speex wide band: 30 kbit/s payload, 16 kHz sampling rate
- OpusOpus (codec)Opus is a low-delay wideband codec intended for applications such as VoIP that will eventually be royalty-free. Opus incorporates technology from the speech-oriented SILK codec and the low-latency CELT codec...
: 10-50 kbit/s, up to 48 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...