Arsen, thanks for your response.
Apparently, even before implementing HTTP calls, I run into this problem when trying to start with the
demo-synth
mock-up:
WARNING[14269]: chan_sip.c:7508 sip_write: Asked to transmit frame type slin16, while native formats is (g722) read/write = g722/slin16
This has the effect of not hearing the audio (in zoiper5 softphone), while the rest of the setup (asterisk / unimrcpserver / zoiper5) seems to be working fine, with response status 200 OK, and normal-looking logs (except for the above warning).
Seemingly, that warning indicates that the SIP channel is trying to transmit a frame in the slin16 (16-bit linear PCM) format, but the native format for the channel is g722 (7200 Hz wideband audio).
Now, the only reference to pcm I see in the demo synth plugin code is this one:
char *file_name = apr_psprintf(channel->pool,"demo-%dkHz.pcm",descriptor->sampling_rate/1000);
And the actual demo files are:
$ find -name "*.pcm"
./data/demo-8kHz.pcm
./data/demo-16kHz.pcm
However in Zoiper5 we use the G.722 codec, and I don't see the slin16 codec listed. Normally unimrcp should be able to convert slin16 to/from g722, right? It's working for our ASR but not for TTS.
Greg