#2672 zrtp MitM Audio is garbled

[Anonymous]

Originally created by: kin... (code.google.com)@gmail.com

What steps will reproduce the problem?
1.Dial voicemail/music on hold on freeswitch

What is the expected output? What do you see instead?
Clean audio.  Garbled audio.

What version of the product are you using? On what device / operating
system?
Latest, nexus one, CM7

Please provide any additional information below.

When using inbound-zrtp-passthru on internal.xml in freeswitch, zrtp is passed correctly through csipsimple on person<->person calls.  When calling server-side extensions like voicemail and music on hold, zrtp MitM takes over, which breaks audio on csipsimple.

Thanks

[Anonymous]

Originally posted by: kin... (code.google.com)@gmail.com

I figured out the problem.  It's not zrtp, so forget about that.  It's the implementation of opus from csipsimple.  It's a clockrate mismatch that's causing the screwy audio.  Freeswitch is sending 48khz audio to opus, as it should, but csipsimple is having to downsample it to 16khz.  pcmu/pcma doesn't have this problem because it uses the bitrate it requested (8khz).

[Anonymous]

Originally posted by: peterv... (code.google.com)@gmail.com

Yes, the problem is that CS's implementation of Opus doesn't work well with others (Freeswitch, Jitsi, etc). I also think the problem lies in the clockrate.

I've brought this up before here https://groups.google.com/forum/#!topic/csipsimple-dev/AUJZ5oN40Xc but I'm not sure Regis has the time or inclination to fix it.

[Anonymous]

Originally posted by: kin... (code.google.com)@gmail.com

It definitely is the clockrate downsampling to 16khz.  Thanks for the heads up

[Anonymous]

Originally posted by: r3gis... (code.google.com)@gmail.com

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Mergedinto: 2629
Status: Duplicate