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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.

Audio Codecs - FreeSWITCH - Confluence

    https://freeswitch.org/confluence/display/FREESWITCH/Audio+Codecs
    Transcodable Audio Codecs. The following codecs can be used when setting codec_string and absolute_codec_string. SILK Skype Audio codec. Provided by mod_silk. iLBC@30i - iLBC using mode=30 which will win in all cases. speex@8000h@20i - Speex 8kHz using 20ms ptime. speex@16000h@20i - Speex 16kHz using 20ms ptime. speex@32000h@20i …

webrtc - No have audio/video in freeswitch - Stack Overflow

    https://stackoverflow.com/questions/33520158/no-have-audio-video-in-freeswitch
    Try using <action application="set" data="bypass_media=true"/> before bridge application. "No media handling mode" this will rule out audio dependencies from freeswitch. Once this change reloadxml and make call again to see if audio works. If it does, then there is no issues with client side. (STUN works good.)

No audio on Fusion/Freeswitch 1.6.20, Debian 8, behind ...

    https://www.pbxforums.com/threads/no-audio-on-fusion-freeswitch-1-6-20-debian-8-behind-firewall-hosted-google-cloud.746/
    Hi all Let me say in advance, Thank you, as I have struggled all day trying to get this working. I have deployed the latest release Fusion/Freeswitch - 1.6.20 - to a Google cloud Debian 8 instance. I've created a new SIP profile for my domain [pbx01.mydomain.com], I've …

Media Handling in FreeSWITCH - moythreads.com

    http://www.moythreads.com/congresos/cluecon2011/media-handling-freeswitch-final.pdf
    • FreeSWITCH core requires the media to be in L16 (signed linear, raw digital audio) format for manipulation (mixing, tone detection etc) • Codec modules encode and decode from/to L16 format • Pass-thru codec modules are dummies (mod_g729, mod_g723_1) FreeSWITCH Audio Codecs © 2011 Sangoma Technologies 8/10/2011 6

Codec Support and Configuration - Media Transcoding …

    https://wiki.freepbx.org/display/MTC/Codec+Support+and+Configuration
    FreeSWITCH Codec Configuration. The Sangoma transcoder will perform transcoding for all codecs listed in the codec module configuration file: sangoma_codec.conf.xml By default, the codec module is already pre-configured to perform all codec translations for G729. If you require only g729 translations you do not need to edit any information.

FreeSWITCH audio, file, and stream formats | Mastering ...

    https://subscription.packtpub.com/book/networking_and_servers/9781784398880/5/ch05lvl1sec40/freeswitch-audio-file-and-stream-formats
    FreeSWITCH audio, file, and stream formats. FreeSWITCH is able to interface automatically with a lot of codecs and file/stream formats, and it can translate between them. This means that a CD-like source at 48 khz, 16 bit, stereo and wideband will be decoded, downsampled, truncated, mixed, and then re-encoded to be sent in a G711 call. Keeping with the general FreeSWITCH …

[Freeswitch-users] Sofia late-negotiation on re-INVITE ...

    https://freeswitch-users.freeswitch.narkive.com/hebcqThy/sofia-late-negotiation-on-re-invite-codec-modification
    FS still in the media path but no transcoding applied. INVITE [pcma,g729] ---> FS ---> INVITE [pcma,g729] 200 OK [pcma] <--- FS ---> 200 OK [pcma] If a SIP ep send a re-INVITE for codec modification, sofia negotiate the new codec on that leg without "asking" the remote party. This result in transcoding and since I have mod_g729 only in ...

FreeSWITCH transcoding - Media Transcoding Cards ...

    https://wiki.freepbx.org/display/MTC/FreeSWITCH+transcoding
    It is important to understand that FreeSWITCH understands only Linear codec and every other codec is translated to Linear in order for FreeSWITCH to interpret. So, in the above example, when Endpoint A sends audio to End point B , two transcoding sessions are requested by FreeSWITCH (alaw-> linear and linear-> G729).

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