Closed assegaf closed 1 year ago
Looks like your config is correct. Check with -s option if G729 is included in invite.
yes, I thought so, all is correct in BS, its the IP PBX who doing wrong things. I just know it by now. Oh Why g729 exist in the world full of open source and everybody use it ...
closed
I am able to compile bcg729 in libbaresip module g729,
but I am not able to compile support g729 with bcg729 in asterisk ip pbx server. I choose bcg729 because the linux server is ARMV8 machine.
I have done the setup of G729 using BG729 and your library. But I search nowhere for defining the account file for accepting G729. currently I use audio_codecs="G722/16000/1,PCMU/8000/1,G729/8000/1" in config file;
and in account fine I have defined 3 codecs,
module g711.so module g722.so module g729.so
and I see BSLibe log (android), I dont see "module: g729 loaded " line log.
12-28 18:04:26.256 4382 4857 D BSLib : aucodec: PCMU/8000/1 12-28 18:04:26.257 4382 4857 D BSLib : aucodec: PCMA/8000/1 12-28 18:04:26.257 4382 4857 D BSLib : module: pcmu loaded, pcma loaded 12-28 18:04:26.257 4382 4857 D BSLib : aucodec: G722/16000/1 12-28 18:04:26.257 4382 4857 D BSLib : module: g722 loaded 12-28 18:04:26.257 4382 4857 D BSLib : aucodec: G729/8000/1
G722 and PCMU is working fine, except G729 still being rejected / transcoded by sip/media server.
have I wrote correct configuration file in account/config file ?
when I force to G729 only in sip server /ippbx server.
12-28 18:16:39.845 4382 4855 I Baresip Service: got uaEvent call closed,No audio codecs/sip:324647716945@192.168.20.XXX/2377312776 AoR sip:324647716945@192.168.20.XXX call 2377312776 that is will be closed is not found,
It seem my audio codec not recognized by libbaresip-android,.