Search Results for "codecs"

jfPhone is a VoIP/SIP SoftPhone for Desktops.

...It provides enterprise-grade VoIP communication with a focus on performance, security, and extensibility. Features Ultra-Fast: Startup in <0.5s, minimal CPU usage HD Audio: Opus and G.711 codecs with adaptive jitter buffer Security: TLS transport, SDES-SRTP encryption, Windows DPAPI credential protection Multi-Account: Up to 5 simultaneous SIP accounts Plugin SDK: C# SDK for custom integrations (CRM, call recording, automation) Plugin Marketplace: Community and premium extensions Headset Support: Jabra, Poly, Sennheiser, Logitech with HID controls Call Quality: Real-time MOS Score, Jitter, Packet Loss and RTT monitoring Enterprise: Silent MSI installer, GPO support, JSON centralized config Compatibility: 3CX, Asterisk, FreePBX, OVH, Twilio, RingCentral, and more Language packages : https://sipline.feelautom.fr/languages

HTA (HTML Application) GUI which offers a very easy way to control your video conferencing codecs via API. Supports: Cisco SX10, SX20, Spark Room 55, Spark Room Kit Plus Avaya/Radvision Scopia XT1000, XT1200, XT5000, XT4000 Yealink VC500, VC800, VC880 For Avaya/Radvision and Yealink you must install the OSWinsck.Winsock ActiveX first. (is included in the package.) This is required because their API interfaces are TCP Socket base.

Intel Integrated Performace Primitives audio/video codecs plug-in for the OPAL/OpenH323 library including G.728, G.729, G.723.1, G.722.2 GSM-FR, GSM-AMR, H.261, H.263, H.264 and MPEG4 part 2.

DOWNLOAD | EXTRACT | USE A ready to use Virtual Image or VMDK of Asterisk 11/13, FreePBX 2.11/12.0. Network: After booting, run #ifconfig -a Then either #dhclient ethX or edit manually /etc/network/interfaces #ifup ethX More details can be found here: Asterisk 11: http://highsecurity.blogspot.com/2013/04/sanjay-asterisk-debian-based-asterisk.html Asterisk 13:...

This ASN.1 compiler turns ASN.1 specifications into C code. The asn1c is shipped together with conformant BER/DER/XER/PER codecs. The X.509, GSM TAP3, MEGACO, RRC and LDAP encoding and decoding examples are part of the source code distribution. NOTE: THE asn1c PROJECT HAS LARGELY MOVED TO GITHUB: http://github.com/vlm/asn1c

Fluent is a suite of tools for the management of diverse VoIP networks. Fluent RTE, a reporting and templating engine, is at the core. Add-on modules are available to manage provisioning, CDR, feature sets, routing and more.

Among the functionality: Include a large variety of codecs (G711, GSM, and SPEEX) - Protocol SIP - Other technical functionalities the support of DTMF (tonalities) although support ENUM (to employ numbers of SIP instead of the addresses of SIP).

VVOIP is voice and video over IP is a P2P application without any protocol like SIP (no server is required) etc.. application. It use decent codecs for video and voice encrypted streams over IP. Moreover it is also comptiable with VoIP modems, you'd con

MiGBase64 is a very fast and small Base64 Codec written in Java. It doesn't create temporary arrays which many other codecs do. 7-100x faster than Sun's and 2-5x faster than Apache Commons codecs. Se homepage for performance charts.

Cross Platform GUI front end for the Hawkvoice Codecs, Written in wxPython. Primary intended use is for low bandwidth speech such as HamRadio or AM broadcast. Server/Client style. for Win32/OSX/Linux platforms.