[Freeswitch-users] Opus problem

Tue Sep 26 08:18:30 UTC 2017

Hi,

I have solved the issue. It is caused by the variable I set before

execute_on_answer=spandsp_start_fax_detect transfer '*11 XML default’ 10

Somehow the bug placed by this directive effects the decoder when the codec is opus. When 10 seconds expire, it turns back to normal.

Thanks, 

/Volkan

> On 25 Sep 2017, at 22:46, Volkan Oransoy <voransoy at gmail.com> wrote:
> 
> Michael,
> 
> Thank you for your reply. I have removed that directive and it is still the same. 
> 
> In fact I started with the example config at https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+And+The+Opus+Audio+Codec <https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+And+The+Opus+Audio+Codec> and I tried different directives like asymmetric sample rate which may effect this issue. Right now my current config is the same as the reference document, but the issue persists.
> 
> <configuration name="opus.conf">
> <settings>
> <param name="use-vbr" value="1"/>
> <param name="use-dtx" value="0"/>
> <param name="complexity" value="10"/>
> <param name="maxaveragebitrate" value="14400"/>
> <param name="maxplaybackrate" value="8000"/>
> <param name="packet-loss-percent" value="15"/>
> <param name="keep-fec-enabled" value="1"/>
> <param name="use-jb-lookahead" value="1"/>
> <param name="advertise-useinbandfec" value="1"/>
> </settings>
> </configuration>
> 
> /Volkan
> 
> 
> 
>> On 25 Sep 2017, at 22:35, Michael Jerris <mike at jerris.com <mailto:mike at jerris.com>> wrote:
>> 
>> This is due to an incomplete implementation of the asymmetric sample rate feature.  It should work fine without that setting.
>> 
>> On Mon, Sep 25, 2017 at 2:22 PM Volkan Oransoy <voransoy at gmail.com <mailto:voransoy at gmail.com>> wrote:
>> Hi All,
>> 
>> I am trying to transcode between OPUS and G711. I encountered a strange problem which I couldn’t find any clue about the cause. After the channel has been answered with opus, I hear a strange sound (highly distorted version of my voice)  for 10 seconds and it starts to operate properly with the message "[DEBUG] switch_core_media_bug.c:1198 Removing BUG from…” . I can reproduce same problem with echo application. Here is my logs and opus.conf.xml files. Did anybody experienced this issue?
>> 
>> Thanks.
>> 
>> /Volkan
>> 
>> 
>> My FreeSWITH Version:
>> FreeSWITCH Version 1.6.16+git~20170407T164913Z~3da6bd0108~64bit (git 3da6bd0 2017-04-07 16:49:13Z 64bit)
>> 
>> Opus.xml config:
>> <configuration name="opus.conf">
>> <settings>
>> <param name="use-vbr" value="1"/>
>> <param name="use-dtx" value="1"/>
>> <param name="complexity" value="10"/>
>> <param name="maxaveragebitrate" value="14400"/>
>> <param name="maxplaybackrate" value="8000"/>
>> <param name="packet-loss-percent" value="15"/>
>> <param name="keep-fec-enabled" value="1"/>
>> <param name="use-jb-lookahead" value="0"/>
>> <param name="advertise-useinbandfec" value="1"/>
>> <param name="asymmetric-sample-rates" value="1"/>
>> </settings>
>> </configuration>
>> 
>> Freeswitch Log:
>> 2017-09-25 22:01:06.277462 [NOTICE] switch_channel.c:1104 New Channel sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> [1048c869-8d71-41fd-b0b7-f8fbd01a0d93]
>> 2017-09-25 22:01:06.277462 [DEBUG] switch_core_state_machine.c:584 (sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com>) Running State Change CS_NEW (Cur 2 Tot 3196652)
>> 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:9828 sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> receiving invite from 192.168.100.2:5060 <http://192.168.100.2:5060/> version: 1.6.16 git 3da6bd0 2017-04-07 16:49:13Z 64bit
>> 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> entering state [received][100]
>> 2017-09-25 22:01:06.277462 [DEBUG] sofia.c:7058 Remote SDP:
>> v=0
>> o=- 8070049943 45777 IN IP4 192.168.100.2
>> s=dvjtiwu
>> c=IN IP4 192.168.100.2
>> t=0 0
>> m=audio 10036 RTP/AVP 103 8 0 9 101
>> a=rtpmap:103 opus/48000/2
>> a=fmtp:103 maxplaybackrate=8000;maxaveragebitrate=15500;useinbandfec=1;usedtx=1
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=ptime:20
>> a=nortpproxy:yes
>> .
>> .
>> .
>> .
>> .
>> 2017-09-25 22:01:06.337394 [INFO] mod_dialplan_xml.c:637 Processing 2223 <2223>->*71 in context default
>> Dialplan: sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> parsing [default->echo] continue=false
>> Dialplan: sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> Regex (PASS) [echo] destination_number(*71) =~ /^\*71$/ break=on-false
>> Dialplan: sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> Action answer()
>> Dialplan: sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> Action echo()
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:286 (sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com>) State Change CS_ROUTING -> CS_EXECUTE
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:643 (sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com>) State ROUTING going to sleep
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:584 (sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com>) Running State Change CS_EXECUTE (Cur 2 Tot 3196652)
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:650 (sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com>) State EXECUTE
>> 2017-09-25 22:01:06.357410 [DEBUG] mod_sofia.c:198 sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> SOFIA EXECUTE
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_state_machine.c:328 sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> Standard EXECUTE
>> EXECUTE sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> answer()
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[opus:116:48000:20:0:1]
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [opus:116:48000:20:0:1] ++++ is saved as a match
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[PCMU:0:8000:20:64000:1]
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [opus:103:48000:20:0:1]/[PCMA:8:8000:20:64000:1]
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[opus:116:48000:20:0:1]
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[opus:116:48000:20:0:1]
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4484 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[opus:116:48000:20:0:1]
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4429 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4345 Set telephone-event payload to 101 at 8000
>> 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps]
>> 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz]
>> 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps]
>> 2017-09-25 22:01:06.357410 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz]
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:3049 Set Codec sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> opus/8000 20 ms 960 samples 0 bits 1 channels
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_codec.c:111 sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> Original read codec set to opus:116
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:4747 sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> Set 2833 dtmf send payload to 101 recv payload to 101
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:6803 AUDIO RTP [sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com>] 192.168.100.12 port 23594 -> 192.168.100.2 port 10036 codec: 103 ms: 20
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_rtp.c:4096 Starting timer [soft] 960 bytes per 20ms
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7109 sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> Set 2833 dtmf send payload to 101
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7116 sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> Set 2833 dtmf receive payload to 101
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:7139 sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> Set rtp dtmf delay to 40
>> 2017-09-25 22:01:06.357410 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com>!
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_channel.c:3473 (sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com>) Callstate Change RINGING -> EARLY
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media.c:6786 Audio params are unchanged for sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com>.
>> 
>> 2017-09-25 22:01:06.357410 [DEBUG] mod_sofia.c:850 Local SDP sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com>:
>> v=0
>> o=FreeSWITCH 1506342472 1506342473 IN IP4 192.168.100.12
>> s=FreeSWITCH
>> c=IN IP4 192.168.100.12
>> t=0 0
>> m=audio 23594 RTP/AVP 103 101
>> a=rtpmap:103 opus/48000/2
>> a=fmtp:103 useinbandfec=1; usedtx=1; maxaveragebitrate=14400; maxplaybackrate=8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=sendrecv
>> 
>> 2017-09-25 22:01:06.357410 [NOTICE] mod_dptools.c:1312 Channel [sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com>] has been answered
>> 2017-09-25 22:01:06.357410 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> entering state [completed][200]
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_core_media_bug.c:828 Attaching BUG to sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com>
>> 2017-09-25 22:01:06.357410 [DEBUG] switch_channel.c:3772 (sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com>) Callstate Change EARLY -> ACTIVE
>> EXECUTE sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> echo()
>> 2017-09-25 22:01:06.737463 [DEBUG] sofia.c:7048 Channel sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com> entering state [ready][200]
>> 2017-09-25 22:01:06.837459 [DEBUG] switch_rtp.c:7229 Correct audio ip/port confirmed.
>> 2017-09-25 22:01:06.837459 [DEBUG] switch_core_io.c:448 Setting BUG Codec opus:116
>> 2017-09-25 22:01:06.837459 [DEBUG] mod_opus.c:596 Opus encoder: set bitrate based on maxaveragebitrate value found in SDP or local config [14400bps]
>> 2017-09-25 22:01:06.837459 [DEBUG] mod_opus.c:613 Opus encoder: set audio bandwidth to [NARROWBAND] based on maxplaybackrate value found in SDP or local config [8000Hz]
>> 2017-09-25 22:01:06.837459 [DEBUG] switch_core_io.c:780 Engaging Read Buffer at 320 bytes vs 15
>> [...CREEPY VOICE STARTS HERE AND ENDS WITH THE BELOW MESSAGE...]
>> 2017-09-25 22:01:16.017463 [DEBUG] switch_core_media_bug.c:1198 Removing BUG from sofia/internalopus/2223 at sip.example.com <mailto:sofia/internalopus/2223 at sip.example.com>
>> 
>> 
>> 
>> _________________________________________________________________________
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