[Freeswitch-dev] FS complained on RTP packet.

[Johny Kadarisman Kwan]

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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