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FreeSWITCH And The Opus Audio Codec - FreeSWITCH - Confluen…
https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+And+The+Opus+Audio+Codec#:~:text=FreeSWITCH%20support%20for%20Opus%20is%20independent%20from%20the,the%20ability%20to%20negotiate%20asymmetry%20of%20audio%20streams.
FreeSWITCH And The Opus Audio Codec - FreeSWITCH - …
https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+And+The+Opus+Audio+Codec
FreeSWITCH support for Opus is independent from the endpoint in use, so it's available for both generic SIP endpoints and verto-based (WebRTC) clients. Besides being a high quality and low latency audio codec, the main features of Opus for VOIP are FEC and the ability to negotiate asymmetry of audio streams.
Voice Quality Issues · Issue #949 · signalwire/freeswitch ...
https://github.com/signalwire/freeswitch/issues/949
Experiencing issues with audio 'clipping' and 'cracking' and just generally dropping out for users whilst running FreeSWITCH 1.10.5-release-17-25569c1631~64bit on Debian 10. I can't replicate this when doing testing in our quiet hours, but when things busy up we definitely get reports of call quality issues affecting a number of different users ...
Configure Freeswitch to have high quality audio Opus codec ...
https://www.freelancer.se/projects/voip/configure-freeswitch-have-high-quality/?ngsw-bypass=&w=f
Configure Freeswitch to have high quality audio Opus codec through WebRTC to be like Jitsi. Hi Fellow VoIP experts, We are working on a leading web-conferencing open source platform for online learning. This platform uses Freeswitch with Opus codec through WebRTC; so directly through the browser.
JitterBuffer - FreeSWITCH - Confluence
https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer
(local sip user) -----> FreeSWITCH -----> (remote gateway) Where the link between FreeSWITCH and the remote gateway has jitter, and say the local SIP user has no jitter buffering on their IP-phone. This will help the voice quality for the incoming audio. Other usage rtp_jitter_buffer_plc
mod_opus - FreeSWITCH - Confluence
https://freeswitch.org/confluence/display/FREESWITCH/mod_opus
FreeSWITCH And The Opus Audio Codec ... The usage of FEC will improve the audio quality because there will be redundant information in the packets eventually arriving at the decoder after the congestion adventure.This is just an example, don't take these values above exactly as they are (60, 70 , 10) . ...
Media Handling in FreeSWITCH
http://www.moythreads.com/congresos/cluecon2011/media-handling-freeswitch-final.pdf
• FreeSWITCH supports a wide range of codecs – Narrowband (G.711, G.726, G.723.1, G.729AB, Speex …) – Wideband (G.722, G.722.1, G.722.2, Speex …) – Ultra-wideband (G.722.1C, Speex) – CD-quality (CELT) • FreeSWITCH core requires the media to be in L16 (signed linear, raw digital audio) format for manipulation (mixing, tone detection etc)
Why is the problem with sound from freeswitch ...
https://helperbyte.com/questions/247121/why-is-the-problem-with-sound-from-freeswitch
Outgoing audio is (cell phone sound comes). However, if asterisk is to connect softfon the sound quality at altitude. In all directions. If you dial directly with asterisk (connected with 105 extensions) that sound all too well. I took a dump on the asterisk sound when calling through the entire chain more normal. For freeswitch already comes ...
Switch core variables - FreeSWITCH - Confluence
https://freeswitch.org/confluence/display/FREESWITCH/Switch+core+variables
When greater than 0, this variable is set to the divisor of the silence generating function. 400 or 1400 are common values set, but you may experiment with other choices to pick one that sounds best. When true, FreeSWITCH will pick a default comfort noise value. When -1, FreeSWITCH will transmit silence without comfort noise. Usage:
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