[Freeswitch-users] ptime and voice/call quality problem

Michael Collins msc at freeswitch.org
Sat May 28 01:02:37 MSD 2011


Any chance you can get a SIP trace of working vs. not working calls? Put
them on pastebin.freeswitch.org and the community will take a look.

-MC

On Fri, May 27, 2011 at 12:31 PM, Michael Gende <mgende at gendesign.com>wrote:

> Hey Folks,
>
> Kind of strange issue here, tied to some other stuff I've posted about.
> Anyway, does anyone know how to set ptime or where that is configured?
>
> Please forgive if this is staring me in the face somewhere, I've not found
> anything satisfactory just looking around. But, I am slow sometimes.
>
> What's up?  We have an FS that has two incoming numbers. One of them, just
> recently, started doing the following (from our VoIP ISP tech guy):
>
> All calls that work (to the second number) network originates the call with
> maxptime=20, we respond with ptime=20 and everything works fine.  The calls
> that do not work (primary number) network originates the call with
> maxptime=30, we respond with ptime=30 then immediately re-invite with
> ptime=20.  I suspect somewhere in this re-invite either your switch or our
> provider’s switch is getting confused and not sending or processing RTP
> based on the re-invite.
>
> Result? The caller can't be heard by the call taker at our FS. Plus, the
> sound quality, even of the ring, is bad.
>
> Sorry so long a post! Right now, we're forwarding the "bad" number to the
> "good". A band-aid. Any ptime experts here? We did not change this
> parameter, but did modify the call_timeout up to 19 from 15 trying to give
> 'em another ring. Then back down once this started to no avail.
>
> Thanks for reading this far if you did. Any advice appreciated.
>
>
>
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