In the fast-paced world where digital communication is key, you need a completely open, royalty-free and extremely versatile audio codec with no incompatibilities and licensing issues.
A codec is necessary for sending and receiving compressed audio files through the internet. It translates audio into compressed digital bits that can be sent through the network – allowing voice and VoIP to function optimally while giving users a richer and more natural sound while using the least amount of bandwidth. Designed for interactive speech and music transmission over the internet, the Opus Codec is an open and versatile audio codec that incorporates technology from Skype’s SILK and Xiph Org’s CELT codec.
What makes Opus better than other codecs?
Unlike some other codecs, such as Speex and Vorbis, Opus is free to use and was specially designed to function efficiently by having a low delay of 5-66.5 ms and the capacity to support high audio quality. It tremendously surpasses the flexibility and voice quality of older type codecs and provides the user with the best audio experience possible. It is easy to use and is highly compatible with a variety of desktop IP phones and soft phones.
In a nutshell, Opus will deliver you from terrible audio quality and voice delays and allow you to have access to high bitrate voice, even over less-than-perfect IP networks.
In low-bandwidth environments, the Opus can accomplish way more than other codecs because of its capacity to drive real-time communications across internet connections that might not previously have supported such communications. The rich and natural translation of sound allows the user to experience the feeling of presently being in the same setting as the speaker at the end of the network. It promotes smooth collaboration for users across different corners of the world while upholding its reputation for highly adaptable, high-quality sound translation. Based on its excellent performance and features, the Opus codec can run just as well on unlicenced frequency fixed wireless broadband and LTE networks.
- Bitrates from 6 kb/s to 510 kb/s
- Sampling rates from 8 kHz (narrowband) to 48 kHz (fullband)
- Frame sizes from 2.5 ms to 60 ms
- Support for both constant bitrate (CBR) and variable bitrate (VBR)
- Audio bandwidth from narrowband to full band
- Support for speech and music
- Support for mono and stereo
- Support for up to 255 channels (multistream frames)
- Dynamically adjustable bitrate, audio bandwidth, and frame size
- Good loss robustness and packet loss concealment (PLC)
- Floating point and fixed-point implementation
How do you use it?
For fibre and wireless ISPs who aim to get ahead of the competition and achieve maximum customer satisfaction, even in areas where network capacity is an issue, Grandstream offers a new range of IP Phones with Opus integration which allows the user to experience rich call quality and maintain effective communication. The GRP series from Grandstream comprise a wide range of IP phones for customers to pick from – each device made to meet the specific communication needs of the customer.
Grandstream IP phones available at MiRO include;
- Grandstream 2-Line Carrier Desk Phone w/o PoE
- Grandstream 3-Line Carrier Desk Phone with PoE
- Grandstream 4-Line Carrier Wi-Fi Desk Phone
- Grandstream 6-Line Carrier Desk Phone with PoE
- Grandstream 10-Line Carrier Wi-Fi Desk Phone
MiRO offers you flexible, high-quality, and cost-effective products, with pre-configuration and training provided. Be ahead of your competitors today and drop by at one of MiRO’s offices or click here to find more about these ground-breaking technologies.