Projects tagged ‘java’ and ‘sip’

SIP Communicator is an audio/video Internet phone and instant messenger that supports some of the most popular instant messaging and telephony protocols such as SIP, Jabber, AIM/ICQ, MSN, Yahoo! Messenger, RSS, IRC and soon others like IAX.

Jain-SIP is a Java API to handle SIP signaling.

Sipgate is a german voice-over-ip provider, who also provides a SMS service. Sipgate SMS Client is a little Java program using the Sipgate API to send SMS to mobiles. To send SMS with Sipgate, you normally have to visit the Sipgate website. ... [More] Sipgate SMS Client waits in your system tray, so it is possible to start writing after one single click. [Less]

IMS-Communicator is an IMS softphone based on the old version of the sip-communicator java project, implemented on top of the JAIN-SIP stack and the Java Media Framework API. It can be used with Open IMS Core (http://www.openimscore.org).

JAIN-SIP Proxy Server with Presence Support. JAIN-SIP IM Client

High-level Java API for easily creating SIP enabled VoIP applications. Suitable for either a desktop (softphone, phone applet, incoming call screener) or server-side (auto attendant, ACD, voicemail) application. Defines a generic media provider ... [More] interface, and includes an implementation of that interface which uses the Java Media Framework. Flibble-VoIP is the underlying engine of the VoxCenter softphone NEWS 6/11/2008 - Flibble-VoIP Beta 2 has been released. In addition to the addition of the roadmap items below, many fixes were made to STUN, SIP Signaling, media, DTMF, and authentication. Tested with Vonage, VoIP.com, Broad Voice, and IdeaSIP. News - 5/26/2007: I've modified the way flibble-voip is being packaged. This should be less confusing now, and all of the source and binary packages are available (on sourceforge). Fixed an issue with authentication challenges on outgoing calls. This means the PlaceCall example, with two way audio, is functional with VoIP providers, such as Vonage. News - 5/5/2007: Tested with a Vonage softphone account. Can receive calls, verified two way audio! Roadmap Phase 0: (completed rev-50) SIP proxy registration (tested with the SipExchange proxy) (completed rev-50) Place a Call (completed rev-50) End a Call (completed rev-50) Blind Transfer Roadmap BETA 1: (completed rev-81) Media Integration (JMF) (completed rev-81) G711 / PCMU codec (completed rev-81) BYE handling (completed rev-83) Answer Call (completed rev-87) STUN probe for Public IP on init (completed rev-100) Media File Streaming (.WAV / .MP3) (completed rev-100) Allow for Media Source change during a call (completed rev-106) Responding to INVITE authentication challenges (completed rev-118) DTMF receiving (completed rev-122) DTMF Sending Roadmap BETA 2: (completed rev-148) Cancel Call (completed rev-148) Reject Call (completed rev-148) Hold / Unhold (completed rev-148) Echo Suppression (completed rev-148) SIP Keepalives Roadmap RELEASE 1.1: Call Recording Voice Recognition Text-To-Speech Roadmap RELEASE 1.2: Mute local audio playout Mute microphone Transferee Support SIP over TCP GSM Audio Codec Video (JPEG, H263) Consultatvie Transfer Presence Instant Messaging (IM / SIMPLE) [Less]