Configure OPUS codec

Asterisk Asterisk Support
1

Hello I`m trying to confgiure codec opos on Asterisk however is not working here is my configuration:

pjsip.conf

[trunk]
type = endpoint
context = gw
disallow = all
allow = opus16,ulaw,g729,alaw
direct_media = no
aors = trunk
rpid_immediate = yes

codecs.conf
[opus16]
type=opus
fec=yes
cbr=yes
dtx=yes
packet_loss=30
max_playback_rate=16000
bitrate=16000
max_bandwidth=medium

When I saw my channel I have the following output:
core show channel PJSIP/trunk-00000000

– General –
Name: PJSIP/trunk-00000004
Type: PJSIP
UniqueID: 1561685131.6
LinkedID: 1561685131.6

Connected Line ID Name: (N/A)

Eff. Connected Line ID Name: (N/A)

   Language: en
      State: Up (6)

NativeFormats: (opus16)
WriteFormat: slin48
ReadFormat: slin48
WriteTranscode: Yes (slin@48000)->(opus@48000)
ReadTranscode: Yes (opus@48000)->(slin@48000)

I do not understand why I have Transcode of slin and why is opus 48K if I`m trying to use 16K. Should I do any other configuration ?

PS: Codec OPUS is installed and both sides support OPUS.

Thanks !

2

Opus always provides/accepts a 48kHz stream, it is the way it works. When encoded it may be a 16kHz stream or less but it still “appears” as a 48kHz stream to the rest of the system.

3

The bridge peer is using slin@4800. That can, for example, happen if you originate a local channel without overriding the default codec. The channel itself is using Opus.

As such you need to provide much mroe complete details of your dialplan (or equivalent) and the other chanels involved.

4

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