[Freeswitch-users] Choppy sound with PCMU

[eaf]

Hi,

I'm trying to migrate from Asterisk to FreeSWITCH (really like the way how
it can be programmed), but ran into one issue with sound quality that I just
cannot workaround by myself. I would describe the sound problem as being
"choppy". From time to time small portions of the other party's voice are
dropped, so the voice kind of stutters. This is not too bad, but is really
noticeable, happens in every call and I don't experience the same with
Asterisk running on the same box. I attached two files: freeswitch.wav and
asterisk.mp3 to illustrate my point.

Issue completely goes away, if I set inbound-proxy-media to true.

The way how I test is to connect SPA-2000 via 10mbps LAN to the box directly
exposed to internet, and then dial a toll-free via FutureNine (a SIP
provider).

The codec in use is PCMU. Can't really try PCMA or anything else with this
provider. Only PCMU. Tried to match ptime of provider (30) with ptime of the
SPA, didn't get any improvement. Tried turning off recording, no change
either.

What puzzles me is that even with greedy codec negotiations and with PCMU on
both sides of  FreeSWITCH, it's still saying that TRANSCODING_NECESSARY. I'm
attaching relevant portion of freeswitch.log to illustrate.

The box isn't particularly fast: Linux (Debian 4), CPU - AMD Geode LX800
with 997 bogomips. 256MB RAM. Only one call in progress, so I hope that it's
not a performance issue.

http://old.nabble.com/file/p26594250/freeswitch.wav freeswitch.wav 

http://old.nabble.com/file/p26594250/asterisk.mp3 asterisk.mp3 

http://old.nabble.com/file/p26594250/freeswitch.log freeswitch.log 

Tried both 1.0.4 and 1.0.5pre5. Same results.

What should I do next? Calls are consistently bad with FreeSWITCH, and
consistently show no glitches with Asterisk.

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