Transcoding ([email protected])->([email protected])->([email protected])

Hello,

My system:
PBX Firmware:
12.7.5-1807-1.sng7

FreePBX 14.0.3.13
Current Asterisk Version: 13.22.0

Please help more experienced users how to set codecs.

Incoming calls to the FreePBX are G.711a
Telephones use G.722

The “core show channel …” command shows something like this
Incoming call:
– General –
Name: PJSIP/incoming-gw-00000111
Type: PJSIP
UniqueID: 1537443591.557
LinkedID: 1537443591.557
Caller ID: 222222222
Caller ID Name: 222222222
Connected Line ID: 555555555
Connected Line ID Name: 555555555
Eff. Connected Line ID: 555555555
Eff. Connected Line ID Name: 555555555
DNID Digits: 555555555
Language: en
State: Up (6)
NativeFormats: (alaw)
WriteFormat: slin16
ReadFormat: slin16
WriteTranscode: Yes ([email protected])->([email protected])->([email protected])
ReadTranscode: Yes ([email protected])->([email protected])->([email protected])
Time to Hangup: 0
Elapsed Time: 0h0m38s

Connection to the phone:
– General –
Name: PJSIP/555555555-00000112
Type: PJSIP
UniqueID: 1537443591.558
LinkedID: 1537443591.557
Caller ID: 555555555
Caller ID Name: 555555555
Connected Line ID: 222222222
Connected Line ID Name: 222222222
Eff. Connected Line ID: 222222222
Eff. Connected Line ID Name: 222222222
DNID Digits: (N/A)
Language: en
State: Up (6)
NativeFormats: (g722)
WriteFormat: slin16
ReadFormat: slin16
WriteTranscode: Yes ([email protected])->([email protected])
ReadTranscode: Yes ([email protected])->([email protected])

Is not too much transcoding?
Maybe phones should also use only G.711a

Transcoding to/from G.722 is not that CPU Intensive - G.729 WAS and therefore was warned against - but going from an HD codec to a Non-HD codec or the other way around is not that hard.

Try and find an ITSP that supports G.722 and then all the transcoding happens on their end - 0% impact on your box.

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