[asterisk-bugs] [JIRA] (ASTERISK-26924) Codec OPUS bad quality

[Kevin Harwell (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26924?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Kevin Harwell closed ASTERISK-26924.
------------------------------------

    Resolution: Suspended

Suspended due to lack of activity. This issue can be reopened or looked into further if/when more feedback is given.

> Codec OPUS bad quality
> ----------------------
>
>                 Key: ASTERISK-26924
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26924
>             Project: Asterisk
>          Issue Type: Improvement
>      Security Level: None
>          Components: Codecs/codec_opus
>         Environment: Ubuntu 16.04 i386/x86_64
>            Reporter: TSAREGORODTSEV Yury
>            Assignee: Kevin Harwell
>         Attachments: kharwell-audio.tar.gz
>
>
> Codec OPUS developed by Digium doesn't provide same good quality as open source version in case of packet losses, delays.
> After comparing recorded degraded media with original sample using PESQ results shows average predicted MOS of Digium Opus almost twice less then open source.
> How to reproduce issue:
> 1. Environment: ubuntu 16.04
> 2. Libopus latest stable release 1.1.4 (compiled from sources).
> 3. Asterisk 13.14.0
> 4. Channel driver: chan_sip
> 5. Configure 2 asterisk hosts, where 1st host origination side, 2nd host termination side.
> 6. Configure dialplan: on 1st host Dialplan with AppMonitor
> {noformat}
> [opus-test]
> exten => _X.,1,Monitor(wav,/records/degraded-ID-${RAND(1,1000)},bi)
> exten => _X.,n,Dial(SIP/asterisk_host2/${EXTEN})
> {noformat}
> 7. Configure dialplan: on 2nd host Playback any wave file (8000, 16bit):
> {noformat}
> [opus-test]
> exten => _X.,1,Answer()
> exten => _X.,n,Playback(demo-instruct)
> {noformat}
> 8. Opus config on both:
> {noformat}
> [opus]
> type=opus
> packet_loss=40
> max_playback_rate=48000
> bitrate=24000
> cbr=0
> fec=1
> {noformat}
> 9. Compile ITU-T PESQ utility from
> https://github.com/dennisguse/ITU-T_pesq
> 10. Simulate packet loss/delays on 2nd host with
>     tc qdisc add dev eth0 root netem loss 20% delay 100ms 20ms distribution normal
> 11. Compare recorded degraded media (only incoming channel of course) with original:
> ./itu-t-pesq2005 +8000 original.wav recorded.wav
> P.S. Please keep in mind without applying patch ASTERISK-25629 asterisk doesn't ignore lately arrived RTP no matter of enabled, forced jitterbuffer. Which is make quality worst.



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