Voice over Internet Protocol (VoIP), also called IP telephony, is a method and group of technologies for the delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the Internet. VoIP allows you to make voice calls using a broadband Internet connection instead of a regular (or analog) phone line.

VoIP services convert your voice into a digital signal that goes over the Internet. They transmit media streams using special media delivery protocols that encode audio and video with audio codecs and video codecs. Various codecs optimize the media stream based on application requirements and network bandwidth.  Some implementations rely on narrowband and compressed speech, while others support high-fidelity stereo codecs. The most widely used speech coding standards in VoIP are based on the linear predictive coding (LPC) and modified discrete cosine transform (MDCT) compression methods. Popular codecs include the MDCT-based AAC-LD (used in FaceTime), the LPC/MDCT-based Opus (used in WhatsApp), the LPC-based SILK (used in Skype), μ-law and A-law versions of G.711, G.722, and an open source voice codec known as iLBC, a codec that uses only 8 kbit/s each way called G.729.

Tonmind IP Speaker supports SIP Protocol. It can be compatible to IP server PBX platforms perfectly including Asterisk, CallWeaver, FreeSwitch, OpenPBX, pbx4Linux, sipwitch, sipX, YATE. It also support RTP Multicasting, thus can work with Axis software. Their supporting codecs including various OPUS, G711U, G711A, G722, GSM, MP1, MP2, MP3, WAV, LPCM s16le. 48K OPUS Audio Codec enables excellent sound quality.

Tonmind SIP Speakers Connecting to VoIP Diagram