[Freeswitch-dev] FS complained on RTP packet.

Steven Ayre steveayre at gmail.com
Thu Aug 26 01:13:08 PDT 2010

Which is 40ms... which is what FS tells you. FS will look at the amount
*actually* received, instead of the amount the client claimed it will sent.
It auto-adjusts to that amount, and displays the warning message because
your client is either misconfigured or broken (some SIP phone
implementations lie).

Because it auto-adjusts it shouldn't be an issue, but it could also affect
quality so a warning is given just in case so that you can fix the problem
on the client.


On 26 August 2010 04:30, Michael Jerris <mike at jerris.com> wrote:

> 20 ms of ulaw would be 160 bytes, not 320.
> Mike
> On Aug 25, 2010, at 4:46 PM, Johny Kadarisman Kwan wrote:
> I'm trying to convert a proprietary audio stream into sip/rtp compatible.
> At this point, i'm able to pass sip negotiation with FS, and trying to
> stream PCMU/8000 codec. But something not right on my end, and FS complained
> about the ptime settings.
> I'm sending 320 bytes ulaw payload on every rtp packet. Is that the right
> amount? appreciate if anyone could point some info on these topics.
> Thanks,
> JK
> ===============
> *2010-08-25 16:36:58.142328 [WARNING] mod_sofia.c:1013 We were told to use
> ptime 20 but what they meant to say was 40*
> *This issue has so far been identified to happen on the following broken
> platforms/devices:*
> *Linksys/Sipura aka Cisco*
> *ShoreTel*
> *Sonus/L3*
> *We will try to fix it but some of the devices on this list are so broken,
> *
> *who knows what will happen..*
> *===================*
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