Asterisk and G726 codec with older voip phones

[Vesa]

I hit my head to old DECT handset. Type is Gigaset A12, mostly sold in Europe as part of Gigaset A120 base station. I have Fritz!box for DECT base station use and it support A12 directly. Only issue is codec, A12 support only G726. Fritz!box can do tranlation to alaw but Asterisk support also G726 directly. SIP endpoint setting with G726 support and endresult is totally broken audio. G726 has issues on early voip phones and it seems that DECT phones can have same issue, bit order for audio data is reversed. There is just a few forum post about this issues but in Asterisk pjsip channel documentation solves this issue. Proper endpoint codec config is this:

Code Select Expand

disallow=all
allow=g726
allow=g726aal2
g726_non_standard=yes

Proper codec for "broken" G726 phones is g726aal2 but those phones signals g726. With g726_non_standard Asterisk use g726aal2 when phone states g726. In this case Fritz!box is just pass-through device and A12 handset is the broken device. With this settings A12 has clear and good audio on both direction. Also asterisk offer direct conversion from g726 to g722 or other formats so there is less audio artifact when comparing to g726-alaw-g722 conversion. My phones use mostly G722 HD audio codec, also for analog phones.