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

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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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