Always transcoding to slin narrowband when calling into conference

Asterisk Asterisk Support
1

When dialing into a conference using the Confbridge application, Asterisk is always transcoding to Slin@8000.
Why would that be?

This is Asterisk 13.18.3 on FreePBX distro (v13)

-- General --
           Name: SIP/5682-00001155
           Type: SIP
       UniqueID: 1558040189.59920
       LinkedID: 1558040189.59920
      Caller ID: 5682
 Caller ID Name: Linphone
Connected Line ID: (N/A)
Connected Line ID Name: (N/A)
Eff. Connected Line ID: (N/A)
Eff. Connected Line ID Name: (N/A)
    DNID Digits: 5914
       Language: en
          State: Up (6)
  NativeFormats: (ulaw)
    WriteFormat: slin
     ReadFormat: slin
 WriteTranscode: Yes (slin@8000)->(ulaw@8000)
  ReadTranscode: Yes (ulaw@8000)->(slin@8000)
 Time to Hangup: 0
   Elapsed Time: 0h0m12s
      Bridge ID: 3f0b0324-efc1-40de-8fc9-6eed3bde9ea9
 --   PBX   --
        Context: from-internal
      Extension: STARTMEETME
       Priority: 5
     Call Group: 0
   Pickup Group: 0
    Application: ConfBridge
           Data: 5914,,,
 Call Identifer: [C-00000b40]
      Variables:
BRIDGEPEER=CBAnn/5914-00000705;2
DB_RESULT=
GOSUB_RETVAL=
CALLFILENAME=5914-5914-never-20190516-165630-1558040189.59920
REC_POLICY_MODE_SAVE=
MON_FMT=wav
FROMEXTEN=5682
TIMESTR=20190516-165630
YEAR=2019
MONTH=05
DAY=16
NOW=1558040190
REC_STATUS=INITIALIZED
MAX_PARTICIPANTS=0
MEETME_MUSIC=
MEETME_ROOMNUM=5914
MACRO_DEPTH=0
TTL=64
CALLEE_ACCOUNCODE=
DIAL_OPTIONS=tr
AMPUSERCID=5682
AMPUSERCIDNAME=Linphone 
AMPUSER=5682
REALCALLERIDNUM=5682
TOUCH_MONITOR=1558040189.59920
SIPCALLID=PAClpwpyA4
SIPDOMAIN=10.1.1.178
SIPURI=sip:5682@192.168.1.254
  CDR Variables:
level 1: calledsubaddr=
level 1: callingsubaddr=
level 1: dnid=5914
level 1: cnum=5682
level 1: cnam=Linphone 
level 1: clid="Linphone " <5682>
level 1: src=5682
level 1: dst=STARTMEETME
level 1: dcontext=from-internal
level 1: channel=SIP/5682-00001155
level 1: lastapp=ConfBridge
level 1: lastdata=5914,,,
level 1: start=1558040189.304192
level 1: answer=1558040189.310634
level 1: end=1558040191.162327
level 1: duration=1
level 1: billsec=1
level 1: disposition=8
level 1: amaflags=3
level 1: uniqueid=1558040189.59920
level 1: linkedid=1558040189.59920
level 1: sequence=10725
2

I think the answer is becuase you are using Asterisk 13.

It obviously has to transcode to linear in order to be able to add samples together. Traditinoally that has been 8kHz linear., but I seem to remember examples on the forum using 48kHz, linear, so I suspect there has been a recent change in policy.

3

It’s based on the sample rate of the source codec. In this case it’s ulaw, so it would be 8kHz.

4

When I am using Opus 48kHz, it’s also transcoding to 8kHz slin:

-- General --
           Name: SIP/5682-00001439
           Type: SIP
       UniqueID: 1558112296.69765
       LinkedID: 1558112296.69765
      Caller ID: 5682
 Caller ID Name: Linphone
Connected Line ID: (N/A)
Connected Line ID Name: (N/A)
Eff. Connected Line ID: (N/A)
Eff. Connected Line ID Name: (N/A)
    DNID Digits: 5914
       Language: en
          State: Up (6)
  NativeFormats: (opus)
    WriteFormat: slin
     ReadFormat: slin
 WriteTranscode: Yes (slin@8000)->(slin@48000)->(opus@48000)
  ReadTranscode: Yes (opus@48000)->(slin@48000)->(slin@8000)
 Time to Hangup: 0
   Elapsed Time: 0h0m11s
      Bridge ID: 8823d89a-d302-47ae-82b2-774612d809c9
 --   PBX   --
        Context: from-internal
      Extension: STARTMEETME
       Priority: 5
     Call Group: 0
   Pickup Group: 0
    Application: ConfBridge
           Data: 5914,,,
 Call Identifer: [C-00000d00]
      Variables:
BRIDGEPEER=CBAnn/5914-00000838;2
DB_RESULT=
GOSUB_RETVAL=
CALLFILENAME=5914-5914-never-20190517-125818-1558112296.69765
REC_POLICY_MODE_SAVE=
MON_FMT=wav
FROMEXTEN=5682
TIMESTR=20190517-125818
YEAR=2019
MONTH=05
DAY=17
NOW=1558112298
REC_STATUS=INITIALIZED
MAX_PARTICIPANTS=0
MEETME_MUSIC=
MEETME_ROOMNUM=5914
MACRO_DEPTH=0
TTL=64
CALLEE_ACCOUNCODE=
DIAL_OPTIONS=tr
AMPUSERCID=5682
AMPUSERCIDNAME=Linphone
AMPUSER=5682
REALCALLERIDNUM=5682
TOUCH_MONITOR=1558112296.69765
SIPCALLID=rljRKdcJGk
SIPDOMAIN=10.1.1.178
SIPURI=sip:5682@10.124.195.18
  CDR Variables:
level 1: calledsubaddr=
level 1: callingsubaddr=
level 1: dnid=5914
level 1: cnum=5682
level 1: cnam=Linphone 
level 1: clid="Linphone" <5682>
level 1: src=5682
level 1: dst=STARTMEETME
level 1: dcontext=from-internal
level 1: channel=SIP/5682-00001439
level 1: lastapp=ConfBridge
level 1: lastdata=5914,,,
level 1: start=1558112296.446443
level 1: answer=1558112296.452771
level 1: end=1558112298.164995
level 1: duration=1
level 1: billsec=1
level 1: disposition=8
level 1: amaflags=3
level 1: uniqueid=1558112296.69765
level 1: linkedid=1558112296.69765
level 1: sequence=12530
````Preformatted text`
5

Conferencing always requires a decode to some slin and an encode from some slin to some other codec. The mixing occurs in slin.

The particular slin that’s used for the conference is a function of the internal_sample_rate setting for the bridge. If it’s auto, then it’s dependent upon the participants.

6

Thanks.
The internal_sample_rate setting is not set in confbridge.conf (or confbridge_additional.conf in my case).

I have also noticed an error on the CLI, that the Opus decoding buffer is too small.
What does that mean and could this be the reason why my conference decodes Opus 48kHz to slin 8 kHz instead of slin48?

ERROR[139625][C-00000d00]: codec_opus.c:388 opus_dec_decode: Opus: decoding : buffer too small

7

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