Projects tagged ‘rtp’ and ‘sip’

Asterisk is a complete PBX and telephony toolkit in software. It runs on Linux, *BSD, MacOSX, and Solaris. It provides all of the features you would expect from a PBX and more as it enables ... [More] developers to build customized voice applications of many types. Asterisk does voice over IP in many protocols, and can interoperate with almost all standards-based telephony equipment using relatively inexpensive hardware. [Less]

The Sippy RTPproxy is a high-performance software proxy for RTP streams that can work together with SIP Express Router (SER), OpenSER or Sippy B2BUA. Originally created for handling NAT scenarious it ... [More] can also act as a generic media relay as well as gateway RTP sessions between IPv4 and IPv6 networks. RTPproxy was developed by Maxim Sobolev and now is being actively maintained by the Sippy Software, Inc. The RTPproxy supports some advanced features, such as remote control mode, allowing building scalable distributed SIP VoIP networks. The nathelper module included into the SIP Express Router (SER) or OpenSER as well Sippy B2BUA allow using multiple RTPproxy instances running on remote machines for fault-tolerance and load-balancing purposes. [Less]

Gemeinschaft is the open-source PBX based on Asterisk, MySQL, Apache, and PHP, and is designed for high availability and clustering. It provides automatic provisioning for mass deployment, and can ... [More] handle over 10,000 users. Administration is done via shell scripts or a Web GUI. Hot-desking and mobility are supported. German voice prompts are included. There is a standards-compliant Web GUI with internationalization (and German and English translations). Outbound and inbound routing with full PCRE support is included. [Less]

Gemeinschaft (by Amooma) is the open-source PBX based on Asterisk, MySQL, Apache, and PHP, and is designed for high availability and clustering. It provides automatic provisioning for mass deployment ... [More] , and can handle over 10,000 users. Administration is done via shell scripts or a Web GUI. Hot-desking and mobility are supported. German voice prompts are included. There is a standards-compliant Web GUI with internationalization (and German and English translations). Outbound and inbound routing with full PCRE support is included. [Less]

Sippy B2BUA is a RFC3261-compliant SIP Back-to-back user agent (B2BUA). Unlike a SIP proxy server, which only maintains transaction state, the B2BUA maintains complete call state and participates ... [More] in all call requests. For this reason it can perform number of functions that are not possible to implement using SIP proxy, such as for example accurate call accounting, pre-paid rating and billing, fail over call routing etc. Unlike PBX-type solutions such as Asterisk for example, the B2BUA doesn't perform any media relaying or processing, therefore it doesn't introduce any additional packet loss, delay or jitter into the media path. Sippy B2BUA supports RADIUS AAA protocol, which is compatible with Cisco VoIP gateways allowing it to be seamlessly integrated into existing networks. [Less]

Sems is a extensible media server which helps you adding voice services to your VoIP system.

NEHET project is aimed for IP telephony (VoIP) between classical SIP phones and Flash streaming applications. Main goal of the project is to create a gateway for Flash applications, which can ... [More] communicate with SIP phones. The gateway behave as a server for Flash applications (RTMP server) and as SIP client for communication with other SIP clients. In result, the gateway has to be connected to a SIP server as mediator for all of clients (the gateway included). Used protocols: Flash side RTMP SIP phone side SIP SDP RTP [Less]

Oxsipgen is a SIP (and RTP) Load Generator using the SIPfoundry SIP stack. It enables you to generate many SIP calls using pre-defined scenes. It is designed to be cross-platform and has both a GUI and command-line version.

A high-level API to access IP Multimedia Subsystem (IMS) services. This API hides IMS technology details and exposes service-level support to enable easy development of IMS applications. High-level ... [More] support for the IMS registration mechanism Support for co-location of multiple IMS Services Use of IMS service sessions (based on SIP sessions) Use of Java Media Framework for media transmision. Hiding and encapsulating internal protocols managed and used by the IMS protocol stack (and open at the same time for IMS experts). The IMS Services API allows application developers to easily utilize the IMS functionality without requiring knowledge of the underlying protocols and IMS implementation details. As you can read, the abstraction level of this API is based on JSR-281 (designed for Java ME). There are slightly differences due to the fact that JavaSE offers more possibilites (for instance: audio/video captures can be more than one). The API have been packaged at different levels: IMS Core API - Basic IMS API and framework for building communication services. IMS Core XDM - An framework for provisioning of Communication Services (for instance, presence, messaging) IMS Presence API - An high level abstraction API for interacting with a Presence Server (as specified by OMA Presence. IMS Messaging API - High level abstraction for interacting with a Messaging Server (as specified by OMA Messaging). [Less]

Excellent VoIP software for Android (Technologies: SIP, RTP, SRTP & Codec: G729)