Linphone is an open source SIP client for HD voice/video calls, 1-to-1 and group instant messaging, conference calls etc. Available for iOS, Android, Windows, macOS and GNU/Linux.
2.5.2 SIP客户端选择:Twinkle, Linphone, Kphone 1) 较为广泛应用 2) Twinkle不支持视频通话 3) Linphone和Kphone支持视频通话 4) 支持C/C++ 5) 运行平台: Linux 2.6 OpenSIPS的搭建环境 1) OpenSIPS源代码下载: 用svn down下代码 svn cohttps://opensips.svn.sourceforge.net/svnroot/opensips/branches/...
sip:phone is a Unified Communication Client brought to you by Sipwise GmbH, the open source soft-switch vendor. IMPORTANT NOTE: sip:phone requires an existing account on any Sipwise-based system (e.g. a Sipwise sip:provider CE, sip:provider PRO or sip:carrier system on version 3.x or hi...
2.5.2 SIP客户端选择:Twinkle, Linphone, Kphone 1) 较为广泛应用 2) Twinkle不支持视频通话 3) Linphone和Kphone支持视频通话 4) 支持C/C++ 5) 运行平台: Linux 2.6 OpenSIPS的搭建环境 1) OpenSIPS源代码下载: 用svn down下代码 svn cohttps://opensips.svn.sourceforge.net/svnroot/opensips/branches/...
Sipdroid is a free open-source VoIP and SIP client for android devices. Android users can download it fromGoogle Play, orF-droidstores. The app routes calls dialed from your phone's built-in contacts app to VoIP. In settings you can choose when to use VoIP and when to make standard phon...
SFLphone, open-source multiplatform multi-protocol VoIP client Shtoom: SIP softphone in Python, runs on Windows, Mac, Linux SIP Communicator Audio/Video phone and messenger - Multiplatform - Open Source (also supports XMPP, MSN, AIM, Ya...
PhoneAPI fromOpenWengoproject, written in C. It says to provide easy to use high level API for developing SIP audio and video user agents. Ports include Linux, Windows, and MacOS X. Media stack included. (GPL) PJSIP The SIP (and media) stack features in this website. ...
2.5.2 SIP客户端选择:Twinkle, Linphone, Kphone 1) 较为广泛应用 2) Twinkle不支持视频通话 3) Linphone和Kphone支持视频通话 4) 支持C/C++ 5) 运行平台: Linux 2.6 OpenSIPS的搭建环境 1) OpenSIPS源代码下载: 用svn down下代码 svn cohttps://opensips.svn.sourceforge.net/svnroot/opensips/branches...
WebRTC call between browser and SIP softpphone STUN and TURN TURN server for WebRTC – RFC5766-TURN , Coturn, Xirsys , Twillio NAT traversal using STUN and TURN WebRTC security WebRTC Security Architecture Regulatory/Legal Considerations and CALEA with WebRTC development ...
SaraPhone is an open source SIP WebRTC phone, complete with HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Mute, Notifications. SaraPhone is fully integrated with FusionPBX. Based on SIP.js, SaraPhone works with all WebRTC compliant servers: FreeSWITCH, Asterisk, OpenSIPS, Kamailio, ...