I have been working with WebRTC and utilizing the opus codec.  The sound quality on calls is great.  I have been encountering some file issues however and want to get some input on how to best proceed.  When the call is has both ends in Opus, or if Freeswitch is transcoding the call has no issues.  The only issues I encounter are the following.

1. fs_encode fails to create a valid opus file

When fs_encode executes, you do see mod_opus load successfully.  However, the file that is created cannot be read by opus.org utilities, specifically opusinfo.

The command line I use is:

fs_encode -l mod_opus infile.wav outfile.opus

2. Streaming a file that is encoded in opus format fails

If I encode a file using opus.org tools, opusenc, I have a valid opus file and I am able to open it with other audio applications.  If I call into freeswitch and try to stream out that file, there is no error reported in the console logs, but the audio heard is garbled.

Basically the mod_opus appears to work but reading, writing files appears to have an issue.

My questions.

1. Is there anything I might have configured incorrectly?

2. What information can I grab to help troubleshoot this issue?

Jonathan

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