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https://github.com/signalwire/freeswitch.git
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f1bc56d65b44f1d97e1e9c690175fa8757e42c29
addStream() is a deprecated WebRTC API method, and only works via shimming on some platforms. Internally, it is inconsistent in building an offer SDP in the order which FreeSWITCH expects it, which is audio m-lines first. This can result in intermittent m-line ordering errors on the client. Instead addTrack() can be used to force the media tracks to be added to the peer in the order FreeSWITCH expects.
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Merge pull request #1735 in FS/freeswitch from ~MITCH.CAPPER/freeswitch:FS-11865-fix_mod_com_729_bin_url_change_pr to master
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Merge pull request #1351 in FS/freeswitch from ~DRAGOS_OANCEA/freeswitch-dragos:feature/FS-10530_mod_opusfile to master
Description
FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a Raspberry PI to a multi-core server, FreeSWITCH can unlock the telecommunications potential of any device.
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