TgCli is a telegram client to automate repetitive tasks.

GNU ZRTP C++ provides a library that adds ZRTP support to the GNU ccRTP stack and serves as library for other RTP stacks (PJSIP, GStreamer).

Twinkle is a softphone for your voice over IP and instant messaging communcations using the SIP protocol. You can use it for direct IP phone to IP phone communication or in a network using a SIP proxy to route your calls and messages.

xgoldmon is an utility that converts the USB logging mode messages that various Intel/Infineon XGold modems send to the USB port to gsmtap. It then then sends them to a given IP address to enable users to view cellular network protocol traces in Wireshark.

It supports the following smartphones:

• Samsung Galaxy S4, GT-I9500 variant

• Samsung Galaxy SIII, GT-I9300 variant

• Samsung Galaxy Nexus, GT-I9250 variant

• Samsung Galaxy SII, GT-I9100 variant

• Samsung Galaxy Note II, GT-N7100 variant

GNU oSIP is an implementation of the SIP protocol. It is used to provide multimedia and telecom software developers with an interface to initiate and control SIP sessions.

SpanDSP is a library of DSP functions for telephony, in the 8000 sample per second world of E1s, T1s, and higher order PCM channels. It contains low level functions, such as basic filters. It also contains higher level functions, such as cadenced supervisory tone detection, and a complete software FAX machine.

Libosmocore includes several libraries:

• libosmocore: general-purpose functions

• libosmovty: interactive VTY command-line interface

• libosmogsm: definitions and helper code related to GSM protocols

• libosmoctrl: shared implementation of the Osmocom control interface

• libosmogb: implementation of the Gb interface with its NS/BSSGP protocols

• libosmocodec: implementation of GSM voice codecs

• libosmocoding: implementation of GSM 05.03 burst transcoding functions

• libosmosim: infrastructure to interface with SIM/UICC/USIM cards

LibIAX2 implements the Inter-Asterisk-Protocol for relaying Voice-over-IP (VoIP) communications.

Mumble is an low-latency, high quality voice chat software primarily intended for use while gaming. Mumble consists of two applications for separate usage: mumble for the client, and murmur for the server.

Libre is a portable and generic library for real-time communications with async IO support and a complete SIP stack with support for protocols such as SDP, RTP/RTCP, STUN/TURN/ICE, BFCP, HTTP and DNS.

GNU SIP Witch is a peer-to-peer Voice-over-IP server that uses the SIP protocol. Calls can be made from behind NAT firewalls and without the need for a service provider. Its peer-to-peer design ensures that there is no central point for media intercept or capture and thus it can be used to construct a secure telephone system that operates over the public internet.

Sofia-SIP is a SIP User-Agent library, compliant with the IETF RFC3261 specification. It can be used as a building block for SIP client software foruses such as VoIP, IM, and many other real-time and person-to-person communication services.

Phonesim is a modem emulator that oFono uses for development and testing. This allows oFono to be used by any host without requiring special GSM (or other) hardware.

GNU ccRTP is an implementation of RTP, the real-time transport protocol from the IETF. It is suitable both for high capacity servers and personal client applications. It is flexible in its design, allowing it to function as a framework for the framework, rather than just being a packet-manipulation library.

This package provides an implementation of the Secure Real-time Transport Protocol (SRTP), the Universal Security Transform (UST), and a supporting cryptographic kernel.

A collection of libraries and header files for implementing telephony functionality into custom Telegram clients.

Baresip is a portable and modular SIP user agent with support for audio and video, and many IETF standards such as SIP, SDP, RTP/RTCP, STUN, TURN, ICE, and WebRTC.

GNU Common C++ is an portable, optimized class framework for threaded applications, supporting concurrent synchronization, inter-process communications via sockets, and various methods for data handling, such as serialization and XML parsing. It includes the uCommon C++ library, a smaller reimplementation.

PJProject provides an implementation of the Session Initiation Protocol (SIP) and a multimedia framework.

SIPp can be used to test many real SIP equipements like SIP proxies, B2BUAs, SIP media servers, SIP/x gateways, and SIP PBXes. It is also very useful to emulate thousands of user agents calling your SIP system.

This package provides a VoIP media traffic NAT traversal server and gateway. It implements the STUN (Session Traversal Utilities for NAT) and TURN (Traversal Using Relays around NAT) server protocols.

LibiLBC is a packaging friendly copy of the iLBC codec from the WebRTC project. It provides a base for distribution packages and can be used as drop-in replacement for the non-free code from RFC 3591.

Seren is a simple VoIP program based on the Opus codec that allows you to create a voice conference from the terminal, with up to 10 participants, without having to register accounts, exchange emails, or add people to contact lists. All you need to join an existing conference is the host name or IP address of one of the participants.

This package provides callaudiod, a daemon to route audio during phone calls, and a library.