[Freeswitch-users] Choppy sound with PCMU

eaf erandr-junk at usa.net
Tue Dec 1 13:52:03 PST 2009

I should also add, after browsing through some topics here, that my SIP
provider sends 172-byte RTP frames, which is in accordance with ptime:20
that it gives to FreeSWITCH.

eaf wrote:
> 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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