On 10/20/2011 02:31 PM, planetary_io wrote:
Outfuckingstanding. What bit rate are you getting? MP3 is great for
interop for archiving, but as a streaming codec it's total dogshit.
Ha, yes. Using FFMPEG, I am encoding the Asterisk output to a 32kb
bitrate, 8khz sample rate, 16-bit mono.
You want something like QCELP or AMR-NB if any streaming protocol.
AMR-NB would be interesting, and possible I think. I know FFMPEG can
Optimally an uncompressed / unlicensed storage container & codec too,
but we have to prioritize Interop for the People.
I can archive in OGG no problem, as well. I can provide an OGG stream
even, but most phones don't support that.
We'd like to set up a trial using E2E PSTN & terminal native dialer
Okay. My inbound channels currently come from DIDWW.com. $5 setup, $5
month for 2 simult inbound. You can add more inbound at more cost per.
Trust us when we say that the codec efficiency & noise cancellation
you get from the OEM image/chipset integration murders any over the
top VoIP solution.
We're looking into some SIP gateway providers but the much preferred
path - getting PSTN HW interfaces up on the Asterisk server seems
prohibitive from a cost & governance perspective. That's why we were
Yeah... agreed. For inbound, DIDWW does a great job and cheap. I have
been using them for about two years off and on. I have not used them at
volume, only for a few channels at once.
Have you evaluated any indie teleconf services or SIP gateways? Our
theoretical knowledge exceeds our knowledge of the commercial options.
Not sure how indie you mean, but DIDW and DIDW are low cost, on-demand
There are some more radical telecom folks within the indy* world, that I
can track down and see if they have more thoughts. Mostly once you
interface into the PSTN world, you don't have any options but dealing
with some sort of corporate entity.