No need to worry about "L16" codec - that's talking about signed linear 16 bit audio, i.e. uncompressed, raw audio. If you're talking directly to FreeSWITCH, for example when you call voicemail or call an IVR, then the FreeSWITCH side will definitely be "L16". However, if you are connecting through to another device, like another telephone, then you'll see codec negotiation.
Remember: when audio passes through FreeSWITCH it always gets decoded (unless you're using proxy media, but don't do that) and re-encoded, so you'll always see references to L16 which is the "decoded" audio. The primary reason for decoding everything is that FreeSWITCH can then do fun stuff like eavesdropping, mixing audio, conference calls, etc. You can't (easily) mix encoded audio.
So... your issue is most likely not with a codec mismatch but rather timing on the VM. I'll defer to those who know more about VMs and such.
-MC
FS last git
fusionpbx
===========
Hi list,
I'm testing FS w/ twinkle, put iLBC as first CODEC and don't follow
far end CODEC preference. All extensions have proxy_media set.
When I call the VM (*97) I get a very choppy audio.
FS log says: [DEBUG] switch_ivr_play_say.c:1306 Codec Activated L16@8000hz 1 channels 30ms
when twinkle says: ilbc
I guess that the source of my problem, but how can I change that for
FS to switch to iLBC? (I already modified CODECs names & orders into
conf/vars.xml to set iLBX@30i first, and twinkle is also set to 30ms)
Jean-Yves
--
Question authority.
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