Fails to establish audio channel on ppc64le

[madscientist159]

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Description

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Using Jigasi as a SIP bridge on ppc64le, the application loads and connects (SIP dial-in establishes a call and the dial-in user appears in the conference) but no audio is received either direction.

Current behavior

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Jigasi loads without errors. When dialing in over SIP, a call is established and the dial-in user appears in the conference, but no audio is received either direction.

Jigasi error logs show:

SEVERE net.sf.fmj.media.Log.error():
Failed to build a graph for the given custom options.
Failed to realize: net.sf.fmj.media.ProcessEngine@52a5a34b
  Cannot build a flow graph with the customized options: 
    Unable to transcode format: LINEAR, 48000.0 Hz, 16-bit, Mono, LittleEndian, Signed
      to: opus/rtp, 48000.0 Hz, Stereo
      outputting to: raw.rtp
    Unable to add customed codecs:·
      org.jitsi.impl.neomedia.audiolevel.AudioLevelEffect2@2e8816de
Error: Unable to realize net.sf.fmj.media.ProcessEngine@52a5a34b

Expected Behavior

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The dial-in user can hear other participants and speak to them.

Possible Solution

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I traced this fault to missing native libraries in libjitsi; specifically, if libjnopus.so is missing then the error above will occur. The following native libraries are required to make Jigasi work correctly:

libjnopus.so
libjnportaudio.so
libjnsctp.so

After building them and placing them inside the Jigasi JAR file, SIP dial-in works perfectly. As the fix presumably needs to be applied to libjitsi I'll open a PR there for the ppc64le fixups.

Steps to reproduce

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Attempt to install Jitsi on an OpenPOWER (ppc64le) server.