Search results
Results From The WOW.Com Content Network
VoLTE calling. Voice over Long-Term Evolution (acronym VoLTE) is an LTE high-speed wireless communication standard for voice calls and SMS using mobile phones and data terminals. [1] [2] VoLTE has up to three times more voice [3] and data capacity than older 3G UMTS and up to six times more than 2G GSM.
Enhanced Voice Services (EVS) is a superwideband speech audio coding standard that was developed for VoLTE and VoNR.It offers up to 20 kHz audio bandwidth and has high robustness to delay jitter and packet losses due to its channel aware coding [1] and improved packet loss concealment. [2]
ViLTE, an acronym for "Video over LTE", is a conversational (i.e. person to person) video service based on the IP Multimedia Subsystem (IMS) core network like VoLTE. It has specific profiles for the control and VoLTE of the video service and uses LTE as the radio access medium.
To ensure compatibility, 3GPP demands at least AMR-NB codec (narrow band), but the recommended speech codec for VoLTE is Adaptive Multi-Rate Wideband, also known as HD Voice. This codec is mandated in 3GPP networks that support 16 kHz sampling. [110]
For call setup, the body of a SIP message contains a Session Description Protocol (SDP) data unit, which specifies the media format, codec and media communication protocol. Voice and video media streams are typically carried between the terminals using the Real-time Transport Protocol (RTP) or Secure Real-time Transport Protocol (SRTP).
sipXtapi is an open-source solution for VoIP media processing engine supporting wideband and HD voice that provides RTP and codecs through a plugin framework for use with SIP and other VoIP protocols. Skype uses an audio codec called Silk which allows for extremely high-quality audio.
This part of specification also describes the codec down to the bit level, thus enabling the verification of compliance to the part to a high degree of confidence by use of a set of digital test sequences. These test sequences are described in GSM 06.54 and are available on disks. [2] This standard is defined in ETSI ETS 300 726 (GSM 06.60).
The Adaptive Multi-Rate (AMR, AMR-NB or GSM-AMR) audio codec is an audio compression format optimized for speech coding. AMR is a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates ranging from 4.75 to 12.2 kbit/s with toll quality [ 3 ] speech starting at 7.4 kbit/s.